| 00:01.39 | *** join/#asterisk bart_kkk (i=doli@felix.ibt.com.pl) |
| 00:01.42 | *** join/#asterisk RoyK (n=roy@ti211110a081-5006.bb.online.no) |
| 00:03.25 | bart_kkk | is it possible to set a null callerid? so instead of SIP header: From: "num" <sip:xxx@yyy> asterisk uses: From: <sip:xxx@yyy> |
| 00:03.55 | EmleyMoor | bart_kkk: Set(CALLERID(name)="")? |
| 00:03.58 | bart_kkk | i tried different strings but even if its empty 'asterisk' or peer name pops in |
| 00:04.12 | EmleyMoor | Ah, yes, know what you mean now |
| 00:04.25 | EmleyMoor | I don't think it is possible to empty it totally in that case |
| 00:04.48 | bart_kkk | strange thing this field is empty as required when registering :\ |
| 00:05.15 | lmadsen | I don't think it can be NULL |
| 00:05.16 | bart_kkk | i am trying to mimic globe7 softphone and it seems its one of the few things thats left |
| 00:06.05 | bart_kkk | they use 'from' field of value like From: <sip:7771xxx@84.45.70.8> |
| 00:06.43 | EmleyMoor | bart_kkk: Would name same as number do? |
| 00:07.22 | bart_kkk | not sure what u mean EmleyMoor ? |
| 00:07.37 | bart_kkk | callerid(num) set to callerid(name) ? |
| 00:08.11 | EmleyMoor | bart_kkk: Do you need CALLERID(name) empty or would setting CALLERID(name) to CALLERID(num) do@? |
| 00:08.53 | bart_kkk | i believe it has to be empty if at the very end i need 'From: <sip*' no ? |
| 00:09.10 | EmleyMoor | Well, maybe so - but * can't do that |
| 00:09.38 | ZPertee | anyone know how to I can hangup a linksys ata from asterisk. this is my setup pots >> asterisk fxo >> sip ata >> legacy pbx >> phone. The way my setup works is that asterisk is also connected as extension on the legacy pbx. call comes in on asterisk directed to ATA which is connected to legacy pbx and phone is connected to that. Caller wants forwarded to VM I put on hold (legacy pbx) and dial asterisk extension which takes call a |
| 00:09.38 | ZPertee | nd directs to vm. however ata won't hangup any ideas? |
| 00:10.02 | bart_kkk | EmleyMoor i wonder why it insist on setting 'asterisk' if the callerid string is set to "" |
| 00:10.16 | bart_kkk | emleymoor maybe its a one-liner in a source then i could play with it |
| 00:10.31 | *** join/#asterisk SirThomas_Home (n=SirThoma@209.169.199.174) |
| 00:10.33 | EmleyMoor | bart_kkk: I suspect it may well be |
| 00:13.50 | bart_kkk | EmleyMoor would you consider tis inability to set null callerid a feature or a bug ;) |
| 00:18.48 | EmleyMoor | bart_kkk: I would say it's a feature |
| 00:22.53 | bart_kkk | well cant say i share that point of view :D anyway, thanks for your time |
| 00:31.47 | *** join/#asterisk SwK (n=SwK@m635e36d0.tmodns.net) |
| 00:35.42 | *** part/#asterisk SwK (n=SwK@m635e36d0.tmodns.net) |
| 00:39.12 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584069.dsl.bell.ca) |
| 00:46.38 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 00:47.59 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 00:49.39 | *** join/#asterisk jklehm (n=john@67.53.187.205) |
| 00:49.52 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:50.39 | jklehm | what kind of hardware do i need to have it so I can intercept pstn calls filter them based on number and then either ring internal phones or send to voicemail? |
| 00:50.44 | *** join/#asterisk Telamon (n=telamon@blk-89-240-175.eastlink.ca) |
| 00:51.00 | jklehm | using normal internal phones if possible |
| 00:51.33 | jjshoe | jklehm a pc and an analog card with fxo/fxs ports |
| 00:51.42 | jjshoe | I recommend sangoma with hardware echo cancellation |
| 00:52.03 | *** part/#asterisk Telamon (n=telamon@blk-89-240-175.eastlink.ca) |
| 00:56.24 | jklehm | jjshoe: liek an A200? http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/a200.html |
| 00:56.52 | jklehm | any ideas how much they usually cost? |
| 00:58.06 | *** join/#asterisk ReD-MaN (i=root-rox@172-220.static.golden.net) |
| 00:58.24 | joejaxx | wth is asterisk complaining about a wrong password when it is right :P |
| 00:58.35 | jjshoe | jklehm how many inbound lines? how many phones? |
| 00:59.13 | jklehm | jjshoe: heh just a small office here so 2 inbound lines, with maybe 10 internal phones |
| 00:59.36 | jklehm | so many nonesense calls so the filtering to voicemail would help so much |
| 00:59.56 | jklehm | just dont really have a clue about any of this stuff so anything you can tell me is a big help |
| 01:00.34 | jjshoe | jklehm are your phones now office phones? are you set on keeping them? might be cheaper with sip phones |
| 01:01.13 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 01:01.59 | jklehm | jklehm: they are house phones nothing fancy, which is why its a bother when they all ring for every call =P |
| 01:02.27 | jklehm | not necessarily set on keeping them if it would be cheaper to get something different |
| 01:03.59 | jjshoe | probably better off with entry level polycoms to be honest |
| 01:04.22 | jklehm | so I would have a sip card with polycom phones? |
| 01:04.23 | jjshoe | http://www.voipsupply.com/product_info.php?products_id=1484 |
| 01:04.32 | jjshoe | that's what you would need currently |
| 01:04.44 | jjshoe | that leavs you room for expansion |
| 01:04.44 | jjshoe | OR |
| 01:05.06 | jjshoe | http://www.voipsupply.com/product_info.php?products_id=1335 |
| 01:05.37 | jjshoe | I would recommend picking up a 2 port with echo cancelation, and then evaluating some ip phones |
| 01:05.57 | jjshoe | polycom and aastra are great brands |
| 01:06.04 | Katty | hewwo |
| 01:06.37 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:07.22 | jklehm | ip phones are wireless 802.11? |
| 01:08.05 | jjshoe | jklehm no. |
| 01:08.25 | jklehm | so cabling to each desk is needed then |
| 01:08.26 | jjshoe | you can get 802.11 ip phones, but they are generally junk |
| 01:08.31 | jklehm | ok |
| 01:08.40 | jklehm | just trying to see all the cost |
| 01:09.44 | jklehm | i suppose the cabling isnt so bad since not all the comps are wireless anyways humm, thank you very much for your help jjshoe |
| 01:11.33 | joejaxx | lmadsen: have a minute for a pm? |
| 01:12.06 | jjshoe | jklehm most descent phones have cable pass throughts |
| 01:12.18 | *** join/#asterisk adjohn (n=adjohn@i220-108-148-237.s05.a013.ap.plala.or.jp) |
| 01:12.20 | jjshoe | jklehm so if you're computers are connected via ethernet, you just plug the phone into the wall, and your pc into the phone |
| 01:12.56 | jklehm | jjshoe: ahh excellent |
| 01:13.22 | jjshoe | jklehm keep it in mind when ordering phones, cheaper models don't usually do that, for ex, the aastra 9112i does not, but the aastra 9133i does |
| 01:13.56 | jklehm | probably better to get the pass through and save on the mess |
| 01:24.40 | Katty | so. |
| 01:25.05 | Katty | what does one use to clean their wood floors with. |
| 01:26.54 | _ShrikE | Katty: Orange Glo |
| 01:27.14 | Katty | _ShrikE: does it smell... harsh? |
| 01:27.44 | _ShrikE | No its made from orange oil. |
| 01:27.49 | _ShrikE | http://www.orangeglo.com/ |
| 01:27.59 | _ShrikE | unless you are opposed to the smell of citrus |
| 01:28.21 | Katty | hmm. |
| 01:28.46 | Katty | _ShrikE: do you spritz your floor with it? |
| 01:28.50 | Katty | _ShrikE: or dilute it? |
| 01:29.56 | _ShrikE | Just spray it. |
| 01:30.17 | Katty | hmm. |
| 01:30.20 | Katty | that might work then. |
| 01:30.36 | Katty | i steam my floors...instead of 'mopping' them |
| 01:31.09 | *** join/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 01:31.12 | demlak | hi |
| 01:31.14 | Katty | hai |
| 01:32.01 | demlak | i want to try a new dialplan "application".. but currently im out of ideas... |
| 01:32.10 | Katty | make a survey |
| 01:32.20 | Katty | collect inflimation |
| 01:32.28 | Katty | dump it into exim4/mutt |
| 01:32.31 | Katty | or a csv |
| 01:32.39 | Katty | with some neat audio files |
| 01:32.41 | Katty | attached. |
| 01:32.51 | demlak | wait.. im typing =) |
| 01:34.11 | demlak | i want an extension.. when it´s dialed.. asterisk answers, and then i have to type some numbers.. asterisk should dial this number.. and then connect me to this number (sorry.. my english sucks) |
| 01:35.04 | _ShrikE | demlak: Check into the Read() application and pass that on to Dial() |
| 01:35.17 | demlak | 2 isdn lines.. one incomming.. second outgoing.. dial line 1.. * answers.. typing in a new number... * dials this number on line 2... then line 1 will be connected with line 2 |
| 01:35.30 | demlak | i´ll take a look |
| 01:35.35 | demlak | thx |
| 01:36.02 | _ShrikE | np |
| 01:36.08 | demlak | btw.. im using askozia PBX |
| 01:37.15 | Katty | demlak: look at DISA too |
| 01:37.29 | Katty | demlak: DISA will give you the second dial tone when you dial in, so that you can dial out again. |
| 01:37.57 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 01:38.05 | demlak | nice... |
| 01:38.17 | demlak | seems to be the complete application =) |
| 01:38.23 | [hC] | bangs head |
| 01:38.32 | Katty | [hC]: hai |
| 01:38.33 | [hC] | why am i getting double dtmf when im using rfc2833.. argh |
| 01:38.38 | [hC] | hai kat! |
| 01:39.02 | Katty | _ShrikE: do they test Orange Glo on animals? |
| 01:40.44 | *** join/#asterisk BeeBuu (n=beebuu@219.135.41.233) |
| 01:44.59 | demlak | hmm.. DIAS kicks me out of the line, after pressing the first number |
| 01:45.09 | demlak | 1,Answer() |
| 01:45.09 | demlak | 2,DISA(no-password|local) |
| 01:45.09 | demlak | 3,Hangup() |
| 01:45.13 | demlak | right? |
| 01:46.31 | *** join/#asterisk skipper4545 (n=DK@chello084113018116.7.12.vie.surfer.at) |
| 01:47.19 | skipper4545 | anyone heard of the skype<->asterisk channe |
| 01:47.21 | skipper4545 | l |
| 01:47.35 | skipper4545 | how does that work? |
| 01:47.49 | _ShrikE | Katty: That I would not know. I hope not. |
| 01:48.21 | Katty | hrmm. |
| 01:49.51 | skipper4545 | i think they work it with virtual sound devices...can anyone confirm? |
| 01:51.03 | TJNII | They work with hacks, from what I've heard. |
| 01:51.23 | TJNII | Audio links and the like. |
| 01:53.18 | *** join/#asterisk MAzeMiami (i=MazeMiam@98.211.197.119) |
| 01:53.37 | skipper4545 | they dont know the protocol...they just take whats on the sound device |
| 01:53.41 | MAzeMiami | have a couple FreePBX Notices that i can't seem to fix |
| 01:53.42 | MAzeMiami | http://pastebin.ca/961656 |
| 01:53.50 | skipper4545 | so,,, |
| 01:54.03 | skipper4545 | the same thing would work for flash |
| 01:54.09 | skipper4545 | rtmp <-> rtp |
| 01:55.43 | skipper4545 | this would work for asterisk also..have a sip <-> flash convo |
| 01:56.49 | *** join/#asterisk boblutz (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 01:57.11 | boblutz | hello? |
| 01:59.18 | skipper4545 | if you had a virtual display...even a 3g videocall with a sip ua |
| 01:59.33 | boblutz | skipper4545, Is chess-by-phone a bad idea? |
| 02:01.23 | skipper4545 | i have had people play chess over fidonet...how would a more instant chess be a bad idea? |
| 02:01.41 | boblutz | ~fidonet |
| 02:01.41 | jbot | i heard fidonet is an old BBS message network. Thousands of nodes (BBSs) are (or used to be) connected to the net. Messages are posted and passed upstream, new messages are downloaded in the ungodly hours of the night. http://www.fidonet.org/ is *not* very helpful. Ask about a nodelist. |
| 02:01.54 | boblutz | oic |
| 02:04.17 | skipper4545 | and chess...is not a killer application...it is NOT! it is what people do in their spare time |
| 02:05.42 | boblutz | Would it be confusing to enter 2 pairs of DTMF to move? |
| 02:05.59 | drmessano | Chess is the killer app of 1292 |
| 02:06.05 | skipper4545 | lol |
| 02:06.35 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 02:06.50 | skipper4545 | is that the date...amazing |
| 02:07.18 | boblutz | drmessano, you know chess-by-phone would be fun |
| 02:07.22 | drmessano | Just like today we have a myspace top8.. in yesteryear, we had the "Last will and testament" |
| 02:08.01 | drmessano | if you got a share of someones crap when they died, you made their top8 |
| 02:08.07 | skipper4545 | entering the arithmetic notation (or whatever its called) is a bummer |
| 02:08.08 | drmessano | Now, it's just a daily scoreboard |
| 02:08.15 | boblutz | i dont even have a myspace |
| 02:08.27 | drmessano | 1987? |
| 02:08.47 | skipper4545 | i would rather use my cursor function...which is not supported |
| 02:08.49 | boblutz | 21? |
| 02:09.04 | drmessano | That's awesome.. to be that age and not have a myspace.. I almost regret some of the trolling I have done thats been directed towards you. |
| 02:09.15 | drmessano | Ok, I don't regret it, but I respect you more |
| 02:09.46 | boblutz | am i a troll? |
| 02:09.52 | drmessano | No |
| 02:09.56 | Katty | troller. |
| 02:10.03 | Katty | i want some twix. |
| 02:10.17 | drmessano | But you know, to have me tell you "I dont hate you" is as good as it gets |
| 02:10.27 | drmessano | People wait years for that |
| 02:10.30 | boblutz | its pretty good |
| 02:10.32 | drmessano | But you... maybe someday |
| 02:10.40 | boblutz | +2xp |
| 02:11.22 | drmessano | Someone told me once "You are an angry little man" |
| 02:11.27 | drmessano | I said "Stop sucking up" |
| 02:13.10 | drmessano | listens to the silent nod of agreement |
| 02:13.12 | demlak | ok DIAS works.. thx a lot.. and bye =) |
| 02:13.19 | *** part/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 02:13.25 | drmessano | DIAF works too |
| 02:13.42 | skipper4545 | i just got the voice from flash to a asterisk sip client |
| 02:13.56 | Katty | i has TWIX! |
| 02:13.58 | Katty | and milk. |
| 02:14.07 | drmessano | OMG |
| 02:14.17 | drmessano | "have" |
| 02:14.20 | Katty | i knows. |
| 02:14.24 | drmessano | NOOOOOOoooo |
| 02:14.26 | Katty | no |
| 02:14.31 | Katty | has. |
| 02:14.31 | drmessano | No plurals |
| 02:14.32 | Katty | :< |
| 02:14.36 | Nivex | she is teh lolcat |
| 02:14.38 | drmessano | No first person plurals |
| 02:14.42 | Katty | )_= |
| 02:14.44 | Nivex | she can haz twix! |
| 02:14.44 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 02:14.54 | drmessano | My wife was talking like that the other day |
| 02:14.58 | Katty | i am the embodiment of internet pop culture. |
| 02:15.03 | drmessano | So I shoved her in the attic and cut out a hole to feed her through |
| 02:15.05 | Katty | or so i claim. |
| 02:15.13 | Nivex | I catch myself talking like that once in awhile. |
| 02:15.17 | Nivex | :( |
| 02:15.20 | drmessano | It only made things weird from there |
| 02:15.34 | Nivex | drmessano: Ceiling wife is watching you ... ? |
| 02:15.42 | drmessano | meh :( |
| 02:15.50 | skipper4545 | now i need to do some amr2pcm |
| 02:16.43 | *** join/#asterisk kamitodo (n=kmt@pool-141-156-248-50.res.east.verizon.net) |
| 02:17.32 | skipper4545 | ohh here we go rfc 3267 |
| 02:18.45 | Katty | i only type like that. |
| 02:18.50 | Katty | i'm perfectly normal in Real Life |
| 02:19.01 | boblutz | my name isnt really bob or lutz |
| 02:19.18 | kamitodo | Is anyone using Grand Central with sipphone.com for incoming calls? |
| 02:19.22 | skipper4545 | lutz would be a cool name |
| 02:19.33 | Katty | i knew a lutz. |
| 02:19.40 | skipper4545 | i know a lutz |
| 02:19.42 | Katty | her name was kelly |
| 02:19.46 | Katty | or is, i guess |
| 02:19.50 | skipper4545 | he is an asshole |
| 02:20.06 | boblutz | shrugs |
| 02:20.08 | boblutz | its just a name |
| 02:20.20 | boblutz | better then "kumbang" who comes in here a lot |
| 02:20.22 | skipper4545 | the person isnt |
| 02:21.09 | boblutz | chess-by-phone is so do-able in 20 lines or less |
| 02:21.17 | skipper4545 | or really |
| 02:21.25 | boblutz | wait do-able isnt even a word |
| 02:21.37 | skipper4545 | it is with a "-" |
| 02:23.09 | Katty | oh...my |
| 02:23.22 | Katty | ever get that i had too much sweets feeling? |
| 02:24.00 | boblutz | insulin? |
| 02:24.14 | Katty | no idea. |
| 02:24.18 | Katty | but i feel it |
| 02:24.18 | skipper4545 | have you ever had the feeling you wrote a sip stack..but you just cant write a chess application on it...well thats what im feeling now |
| 02:24.41 | boblutz | skipper4545, There are a couple web chess engines out there |
| 02:24.52 | *** join/#asterisk MazeMiami (i=MazeMiam@98.211.197.119) |
| 02:24.58 | MazeMiami | Hello people |
| 02:25.00 | skipper4545 | but not in 20 lines of code |
| 02:25.04 | Katty | wibbles |
| 02:25.05 | Katty | so |
| 02:25.10 | Katty | what counteracts insulin? |
| 02:25.16 | boblutz | oh, i meant 20 lines for me |
| 02:26.13 | boblutz | app_chess |
| 02:26.18 | MazeMiami | Keep getting a Operation not permitted in my FreePBX Notices |
| 02:26.36 | boblutz | MazeMiami, check permissions && ~freepbx |
| 02:26.50 | boblutz | ~freepbx |
| 02:26.51 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:26.52 | drmessano | Yes ~freepbx is your friend |
| 02:27.04 | skipper4545 | bolutz...cmon thats not to be taken serious...in any way or case |
| 02:27.17 | Katty | hmm. freepbx. |
| 02:27.19 | boblutz | skipper4545, the 20 lines or app_chess? |
| 02:27.19 | Katty | hmm. |
| 02:27.25 | Katty | it seems to complicate things, at times. |
| 02:27.30 | Katty | but so does sweets. ugah. |
| 02:27.57 | MazeMiami | what is ~freepbx |
| 02:28.03 | Katty | jbot: freepbx |
| 02:28.03 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:28.04 | boblutz | ~~ |
| 02:28.04 | jbot | Every moment in which I'm called upon is torture. |
| 02:28.09 | kamitodo | Is anyone using Grand Centra? |
| 02:28.14 | kamitodo | Is anyone using Grand Central? |
| 02:28.26 | MazeMiami | i use them |
| 02:28.35 | MazeMiami | used them |
| 02:28.41 | kamitodo | MazeMiami: with gizmo, I suppose? |
| 02:28.44 | skipper4545 | there is no abstraction for any situation... (like chess) that can be solved in 20 lines of code... period |
| 02:28.46 | kamitodo | used? |
| 02:28.47 | MazeMiami | there no longer accepting nay more memebers |
| 02:28.55 | MazeMiami | any* |
| 02:29.02 | MazeMiami | whats the question |
| 02:29.16 | boblutz | skipper4545, allow me to expand |
| 02:29.33 | skipper4545 | sure....eXpand |
| 02:29.42 | *** join/#asterisk WindBack (n=jorge@host162.190-31-205.telecom.net.ar) |
| 02:30.12 | kamitodo | grand central works perfectly with gizmo but after a while it stops working and I cannot figure out why. if I "reload" it starts working again. |
| 02:30.20 | boblutz | s,n,AGI(my_chess_engine_call_script.sh|${DTMF_INPUT_1}|${DTMF_INPUT_2}) |
| 02:30.25 | kamitodo | "reload" in asterisk that is |
| 02:30.26 | Katty | what is this Grand Centeral thing? |
| 02:30.29 | Katty | jbot: grand central |
| 02:30.33 | Katty | :< |
| 02:30.42 | skipper4545 | did someone say gizmo? what makes that so special? |
| 02:30.50 | Katty | skipper4545: gizmo makes SIP calls |
| 02:30.55 | Katty | skipper4545: it's also a multi IMer. |
| 02:31.02 | Katty | skipper4545: also a pain in the tail. |
| 02:31.05 | skipper4545 | yeah...so does JAJA |
| 02:31.07 | Katty | skipper4545: google gizmo5 |
| 02:31.16 | boblutz | multi IMer like kopete or pidgin? |
| 02:31.30 | Katty | yes. |
| 02:31.35 | Katty | it does aim, yahoo, msn, and myspaceim |
| 02:31.45 | skipper4545 | its just sip |
| 02:31.45 | *** join/#asterisk ta^3 (n=tacvbo@189.146.198.128) |
| 02:31.53 | skipper4545 | xmpp! is the future...i would say |
| 02:31.54 | boblutz | and pidgin cant do sip? ironic? |
| 02:32.02 | Katty | hehe |
| 02:32.08 | kamitodo | the only way to use grand central for incoming calls with asterisk is if you have a gizmo account, register as a user and wait for calls. |
| 02:32.14 | Katty | what is grand central? |
| 02:32.17 | Katty | is it a sip phone? |
| 02:32.23 | skipper4545 | Imm style...at least |
| 02:32.39 | drmessano | Grandcentral is a free number for life from google |
| 02:32.46 | Katty | ah, right. |
| 02:32.55 | drmessano | Theres a web gui where you direct it to your current numbers or a Gizmo account |
| 02:33.21 | Katty | i think i'm just going to wait until smart phones get standardized |
| 02:33.21 | skipper4545 | can you direkt your gizmo to your own proxy? |
| 02:33.26 | Katty | and people aren't afraid to write sip clients |
| 02:33.41 | Katty | skipper4545: the softphone for a desktop will let you connect to a second account |
| 02:33.57 | Katty | skipper4545: the one on my blackberry claimed to do sip calls, but when i called my sip number on asterisk....it didn't work |
| 02:34.02 | Katty | skipper4545: i punted it |
| 02:34.10 | skipper4545 | sorry...the gateway...softphone of the future will let me do anything i want... |
| 02:34.15 | drmessano | No, you just use it in Asterisl |
| 02:34.17 | drmessano | No, you just use it in Asterisk |
| 02:34.33 | drmessano | set up a peer for Gizmo |
| 02:34.48 | Katty | meh. |
| 02:34.50 | Katty | too much trouble. |
| 02:34.56 | drmessano | Grandcentral > Gizmo > Asterisk |
| 02:35.14 | drmessano | Gizmo is good for peer to peer calling.. Tons better than FWD |
| 02:35.16 | kamitodo | the gizmo peer works fine. for a while. then it stops. |
| 02:35.25 | kamitodo | has anyone experienced this? |
| 02:35.26 | drmessano | Mine works fine |
| 02:35.29 | skipper4545 | in any case...i dont like pulver...i dont like gizmo....i like my own phone on my gateway...on my own proxy... |
| 02:35.31 | drmessano | Just need a good config |
| 02:35.43 | mwalling | drmessano: does yours seem to strip dtmf? |
| 02:35.56 | drmessano | Grandcentral strips DTMF |
| 02:36.04 | mwalling | ok, thats what i thought |
| 02:36.19 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.81.24.56) |
| 02:36.26 | drmessano | I give out my GC number as my home number |
| 02:36.34 | drmessano | It rings my gizmo > asterisk and my cell |
| 02:36.40 | drmessano | So its worked out good so far |
| 02:37.22 | skipper4545 | if i would want that stuff i would have chosen skype..seriously...and we all know why it bad in that way |
| 02:37.39 | drmessano | Theres no comparison |
| 02:37.45 | kamitodo | drmessano: mine works. i have it answer the call and dial 1,etc. but after a while (if i don't 'reload') calls go to GC voice mail. e.g. gizmo is not called. |
| 02:37.48 | drmessano | Skype is closed and inflexible |
| 02:37.59 | skipper4545 | fact |
| 02:38.26 | skipper4545 | i need a way to make sip popular |
| 02:38.31 | Katty | myabe i should just write a sip client for j2me |
| 02:38.36 | Katty | write it and be done with it |
| 02:38.45 | drmessano | yes, I have an externally accessible asterisk box, but I also have spare ATA's for Gizmo and direct to my ITSP as backup |
| 02:38.47 | skipper4545 | and i found one to converge flash video/audio with sip |
| 02:38.56 | skipper4545 | thats one step |
| 02:39.15 | drmessano | SIP <> SIP calling will never be popular |
| 02:39.19 | skipper4545 | i will never doubt sip..in any case....i have writte stacks myself |
| 02:40.10 | drmessano | There will always be some wrapper.. Im shocked SMTP and POP are still around, as they require the user to know way too much about transports |
| 02:40.38 | drmessano | Same goes for SIP |
| 02:40.57 | skipper4545 | whats wrong with smtp pop again? |
| 02:40.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:41.10 | drmessano | The fact that users have to configure something to make it work |
| 02:41.13 | *** join/#asterisk klin3d (i=Bsd@pc-32-231-86-200.cm.vtr.net) |
| 02:41.22 | skipper4545 | ok |
| 02:41.53 | skipper4545 | like imap would take one server address is that what you mean? |
| 02:42.06 | drmessano | Exactly why SIP will be Gizmo, FWD, ITSP <> ITSP calling to save $$ on their end, but never "Oh, heres my SIP address" |
| 02:42.43 | drmessano | No, like I add danny@domain.com to my client and it looks up a DNS record and configures it all or something like that |
| 02:42.44 | skipper4545 | enum? |
| 02:43.04 | drmessano | you still have to create an ENUM record |
| 02:43.08 | skipper4545 | 1 |
| 02:43.16 | drmessano | 1 may as well be 1000 |
| 02:43.23 | skipper4545 | y? |
| 02:43.23 | drmessano | Users are stupid |
| 02:43.28 | drmessano | 2. Users are stupid |
| 02:43.30 | drmessano | 3. Users are stupid |
| 02:43.47 | skipper4545 | well..my daddy is 63...and he understood |
| 02:44.03 | skipper4545 | better than 4-5 addresses |
| 02:44.05 | drmessano | i dont want your one or two examples.. ive been working with users for years.. |
| 02:44.18 | drmessano | users do not know or care, for the most part |
| 02:44.21 | skipper4545 | we are talking convergance .-..in any case |
| 02:44.56 | skipper4545 | i understand |
| 02:44.58 | kamitodo | drmessano: is your asterisk box behind firewall? did you do anything special for gizmo? fwd specific ports, etc... |
| 02:45.28 | drmessano | kamitodo: I created a working peer.. firewall is irrelevant |
| 02:45.29 | kamitodo | i can't figure out what makes my setup not work with gizmo after a while |
| 02:45.47 | skipper4545 | well i guess in the matter of convergence...the best thing is enum |
| 02:46.00 | kamitodo | i do have a peer, even sip show peers and sip show registry report it ok. |
| 02:46.27 | drmessano | kamitodo: a peer is not a peer is not a peer.. just because you've registered and it sorta works, doesnt mean you have a good config |
| 02:47.05 | drmessano | enum is awesome.. but let me give you something to consider |
| 02:47.21 | drmessano | I know one ITSP thats testing direct sip calls to their PSTN numbers |
| 02:47.29 | drmessano | A SIP gateway will save them $$$$ |
| 02:47.44 | drmessano | It also mean they can route calls SIP <> SIP amongst their customers |
| 02:47.55 | drmessano | and save $$$ here and there |
| 02:48.23 | drmessano | Thats where SIP will thrive and exist more and more.. not users becoming aware of SIP and making SIP calls |
| 02:48.50 | skipper4545 | ENUM is just a IN on IP |
| 02:48.51 | kamitodo | drmessano: i can't see any problem with my config. can you share yours? email it to me perhaps? or i can email you mine and just point out the problems? |
| 02:49.02 | drmessano | Just as they dont know or care that they have a VOIP PBX.. they have "cool phones" |
| 02:49.07 | drmessano | If they even care that much |
| 02:49.29 | skipper4545 | true |
| 02:49.54 | drmessano | kamitodo: Give me a little while and I will link you |
| 02:50.56 | skipper4545 | so what is the future of POT? |
| 02:51.09 | drmessano | Being smoked? |
| 02:51.39 | skipper4545 | seriously |
| 02:51.43 | drmessano | POTS |
| 02:51.45 | drmessano | Not POT |
| 02:51.47 | drmessano | POT is good smoke |
| 02:51.54 | skipper4545 | plain ol' telephony |
| 02:52.16 | skipper4545 | i didnt say POTS |
| 02:52.20 | drmessano | I cant say i've ever heard POT.. |
| 02:52.33 | drmessano | Theres no such thing as Plain Old Telephony |
| 02:52.41 | drmessano | Theres the PSTN |
| 02:52.53 | drmessano | Which is becoming a big switched VOIP network already |
| 02:52.59 | skipper4545 | you obviously dont work in that field |
| 02:53.06 | drmessano | Ohhh.. |
| 02:53.14 | skipper4545 | oh |
| 02:53.16 | drmessano | Yeah, because I used POTS and PSTN, and you use POT? |
| 02:53.32 | skipper4545 | no i was asking the "one question" |
| 02:54.07 | drmessano | The "one question"? |
| 02:54.25 | skipper4545 | dude...sorry i offended you seriously |
| 02:54.40 | drmessano | You didnt offend me so much as confused me |
| 02:55.11 | skipper4545 | true...do you still feel you can answer it |
| 02:56.00 | skipper4545 | sorry...im swiss |
| 02:56.23 | drmessano | The future of telephony is VoIP |
| 02:56.44 | skipper4545 | uhnm,,,thanks |
| 02:56.59 | drmessano | POTS, PSTN, POT, or however you want to call it, is going IP |
| 02:57.31 | skipper4545 | what protocol?` |
| 02:57.37 | drmessano | Applications that require POTS lines for legacy equipment will come from an ATA, more or less |
| 02:57.59 | skipper4545 | ata? |
| 02:58.00 | skipper4545 | lol |
| 02:58.02 | drmessano | Who really knows.. A mix, I am sure.. different at different levels |
| 02:58.05 | drmessano | Yes |
| 02:58.29 | skipper4545 | yea...i really like my skinny phone |
| 02:58.36 | drmessano | Blah |
| 02:58.41 | skipper4545 | true |
| 02:58.45 | drmessano | Cisco sux |
| 02:58.49 | skipper4545 | true |
| 02:59.02 | skipper4545 | we had the call server or whatever it was called |
| 02:59.07 | skipper4545 | it SUCKED!!! |
| 02:59.26 | kamitodo | drmessano: link me? thanks! Not quite sure what you meant though. Provide a link? |
| 02:59.30 | skipper4545 | snom rules for example |
| 02:59.39 | drmessano | Yes kamitodo |
| 03:00.08 | skipper4545 | do you use any sbc? |
| 03:00.15 | skipper4545 | in your infrastructure? |
| 03:00.36 | drmessano | Nope |
| 03:00.41 | skipper4545 | k |
| 03:00.46 | kamitodo | drmessano: TIA |
| 03:01.02 | drmessano | kamitodo: PM |
| 03:01.20 | drmessano | Go there, and check what I have posted.. That's as bare a config as you need |
| 03:01.34 | skipper4545 | i like the juniper peerpoint |
| 03:02.25 | drmessano | BRB, got some honey-do's to handle |
| 03:04.10 | *** join/#asterisk RoyK (n=roy@ti211110a081-5006.bb.online.no) |
| 03:08.13 | Katty | dumdedum |
| 03:08.35 | *** join/#asterisk Olobola (i=Olobola@69.sub-70-212-202.myvzw.com) |
| 03:10.18 | skipper4545 | i have written some back2back code for asterisk, where do i submit it? |
| 03:11.50 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 03:12.38 | skipper4545 | hi...i have some code to submit...where do i go about?? |
| 03:18.02 | TJNII | prolly bugs.digium.com |
| 03:28.38 | *** join/#asterisk xanderp (n=pphillip@98.220.163.8) |
| 03:29.53 | xanderp | anyone here use viatalk? wondering if their unlimited plan is really an unlimited* plan with the emphasis on the * small print |
| 03:31.10 | xanderp | that wasn't very clear sarcasm... sorry. I meant to imply that many voip provider's unlimited plans have an asterisk next to them, indicating small print where they define unlimited as like 5000 minutes or some such crap... |
| 03:32.20 | *** part/#asterisk xanderp (n=pphillip@98.220.163.8) |
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| 04:00.22 | lmadsen | chicka chicka |
| 04:01.29 | lmadsen | xanderp: Mix Networks has a truly unlimited plan |
| 04:06.47 | jklehm | can a modem act as fxo? |
| 04:08.19 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 04:09.15 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:12.22 | MAzeMiami | Is there a script to add multiple Extensions to DID's from your DID provider to Free PBX |
| 04:15.42 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.108.27) |
| 04:28.57 | riddlebox | does anyone know the Avaya Legend/Magix system at all? |
| 04:29.27 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 04:34.42 | lmadsen | MAzeMiami: that's the kind of stuff you have to program yourself |
| 04:34.58 | MAzeMiami | i tough so ;) |
| 04:35.00 | lmadsen | jklehm: only a certain kind, and it doesn't work very well |
| 04:35.16 | MAzeMiami | was hoping some one may have releaser there own copy |
| 04:35.23 | lmadsen | MAzeMiami: every situation is different. Learn some PHP because it isn't very hard, and will do all of that kind of stuff |
| 04:35.39 | MAzeMiami | yep thanks! |
| 04:36.10 | lmadsen | MAzeMiami: if you know some basic PHP, that kind of stuff isn't very hard. You have to learn some programming fundamentals, then things like that, and asterisk dialplan, become a lot easier |
| 04:36.43 | MAzeMiami | yea i forever have to hire php progrmaers for silly stuf like this |
| 04:36.57 | *** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com) |
| 04:37.01 | MAzeMiami | i need to learn it |
| 04:37.16 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:37.30 | lmadsen | it's not really that hard. I'm not a programmer, but being able to create simple scripts isn't that hard if you learn even just a few basics |
| 04:37.39 | lmadsen | it's about training your brain to think in that manner |
| 04:37.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:37.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:38.10 | lmadsen | now the epitome of thinking like that has just entered the room :) |
| 04:38.30 | MAzeMiami | yea it would be a simple script scrap 2 fileds and then fill in a couple |
| 04:39.10 | lmadsen | just import a csv file into an array and parse through the array within' a loop to write the required entries in the format you wish |
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| 04:42.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 04:45.47 | *** part/#asterisk DoDaT69 (n=DoDaT69@internal.digitalson.com) |
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| 05:05.20 | Juggie | lmadsen, what happened to your visitor |
| 05:05.34 | lmadsen | nothing unfortunately |
| 05:05.56 | lmadsen | we get along fine, but I think that's about all that's gonna happen |
| 05:06.00 | Juggie | ahhh |
| 05:06.07 | Juggie | another girl on the friend ladder |
| 05:06.16 | lmadsen | indeed |
| 05:06.19 | Juggie | i had to learn to let those ones go |
| 05:06.23 | lmadsen | yeppers |
| 05:06.28 | lmadsen | I know what to look for |
| 05:06.35 | lmadsen | been there done that... I'm no rookie |
| 05:06.37 | Juggie | well you have another chick tomorow :) |
| 05:08.26 | lmadsen | yep, I have a better feeling about that one |
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| 05:50.51 | loompek | exten => 1,1,Dial(SIP/${EXTEN},60,rm) |
| 05:50.59 | loompek | this should generate both dial tone and music on hold.. right? |
| 05:51.36 | jameswf-home | ~ch5 |
| 05:51.36 | jbot | Read about extensions DialPlans etc.. in Chapter 5 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 05:53.25 | russellb | ~ch17 |
| 05:53.34 | russellb | ~ch-3 |
| 05:53.52 | putnopvut | ~chpi |
| 05:53.53 | russellb | ~ch4 |
| 05:53.55 | jameswf-home | eh only ded 5 and 6 |
| 05:53.58 | russellb | ~appendixq |
| 05:53.59 | jameswf-home | *did |
| 05:54.05 | russellb | ~ch6 |
| 05:54.06 | jbot | Read about Advanced extensions DialPlans etc.. in Chapter 6 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 05:57.37 | *** join/#asterisk keith4 (n=kbe2@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 06:03.48 | jameswf-home | heh http://www.tuxparty.com/ |
| 06:04.29 | *** join/#asterisk infamy (n=infamy@216.19.180.122.novuscom.net) |
| 06:05.35 | infamy | I'm looking a linux distro with asterisk designed to run on a cf card. Any recommendations? |
| 06:05.43 | russellb | astlinux |
| 06:06.22 | keith4 | debian |
| 06:06.28 | *** join/#asterisk steliosk (n=Stelios@79.131.104.95) |
| 06:06.59 | infamy | tried astlinux, it does not really have a webgui for config. besides that its really nice. |
| 06:07.12 | infamy | keith4, debian is too large, trying to stay under 512mb |
| 06:07.16 | russellb | you could put the asterisk-gui on there really easily |
| 06:07.20 | keith4 | is not! |
| 06:07.23 | russellb | it's an all client side gui |
| 06:07.37 | keith4 | infamy: i got debian etch onto a 256M cf card |
| 06:08.00 | infamy | hmm debian on 256mb? with asterisk and asterisk gui? |
| 06:08.04 | keith4 | no |
| 06:08.09 | infamy | (what are the ram requirements like) |
| 06:08.15 | keith4 | my debian with asterisk on CF is on a 1G CF |
| 06:08.18 | keith4 | with room to spare |
| 06:08.21 | infamy | russellb: what do you mean its an all client side? |
| 06:08.41 | russellb | it's just html and javascript ... and talks directly to a built-in http server in asterisk |
| 06:08.59 | infamy | keith4: yeah i only have 512mb in this box, also, i have to worry about ram usage, as the box only has 128mb |
| 06:09.02 | russellb | modifies the config files directly ... no db or anything |
| 06:09.34 | infamy | russellb: hmm nice what version of Asterisk does it need? (astlinux is running like 1.2.8 on there latest stable) |
| 06:09.39 | keith4 | infamy: my deb/ast only uses.... *checks free -m* 52M of ram |
| 06:09.45 | russellb | it requires asterisk 1.4 |
| 06:09.48 | keith4 | ... at 2 AM on a saturday |
| 06:10.24 | infamy | k so astlinux is out then.. debian linux with barebones asterisk + asterisk gui is an option |
| 06:10.31 | santoshr | Is it possible to do dial(h323/number@ip) .. or this has to be done via gnugk |
| 06:10.37 | infamy | is there no other distros out there for embedded use? |
| 06:10.48 | keith4 | infamy: a 1G CF card is like $15.... |
| 06:11.12 | infamy | keith4: yeah i know.. but i sort of want this working tonight.. so i have 64mb, or 512mb.. |
| 06:11.24 | keith4 | you could probably do 512, but it'll be tight |
| 06:12.08 | coppice | $15? 1G CF cards come free with corn flakes |
| 06:12.15 | keith4 | santoshr: h323 is a channel, just like any other... see h323.conf |
| 06:12.25 | keith4 | coppice: the "industrial" grade ones are like $15 |
| 06:12.30 | keith4 | whatever that means |
| 06:12.36 | infamy | i'm really surprised that there are no other linux distro out there for embedded asterisk, askozia is really nice but it wont work for me. (i need something that can boot via coreboot) |
| 06:12.51 | coppice | it means the stick a different label on, and double the price |
| 06:12.57 | keith4 | ha |
| 06:12.58 | keith4 | probably |
| 06:13.17 | coppice | to 2 packets of corn flakes |
| 06:13.18 | keith4 | my asterisk box uses a cheapo ebay CF to IDE adapter and a SanDisk camera card |
| 06:13.48 | keith4 | but we've built systems for people using fancy "industrial" CF that plug right into the IDE port, which is nice |
| 06:13.53 | keith4 | much, much lower profile |
| 06:14.18 | keith4 | like this: http://www.emphase.com/fdm4000 |
| 06:17.17 | keith4 | infamy: you might want to read this http://www.usenet-forums.com/linux-general/72638-installing-linux-100-megs-less.html |
| 06:18.54 | the_5th_wheel | for some reason my manager interface wont load. I ahve the config file in place, but r asterisk runnit just doesnt listen on the port |
| 06:19.20 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
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| 06:24.33 | santoshr | keith4 i read the conf setup it up. Bit i am not able to dial. It does do anything. disconnects the call saying |
| 06:24.35 | santoshr | <PROTECTED> |
| 06:25.02 | santoshr | i am able to get a call in using chan_h323 but am not able to dial out |
| 06:26.22 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 06:26.52 | santoshr | am using asterisk 1.4.4 and chan_h323.so that came with it |
| 06:42.27 | santoshr | any ideas guys am i doing something. IS there a requirement like sip.conf to have a peer of the IP the call is being sent to |
| 06:43.54 | infamy | keith4: which debian repo has asterisk 1.4? |
| 06:44.15 | infamy | looks like i can shoehorn this into 512mb |
| 06:45.08 | plik | infamy: build from source for the latest bug-fixes |
| 06:46.15 | infamy | cant really do that.. only have 187mb free... |
| 06:54.22 | infamy | this is going to be tight... debian lenny + asterisk on 512mb cf card.. |
| 06:59.07 | coppice | oh, for the good old days, running Unix on a 256k RAM + 10MB disk machine. :-) |
| 07:04.56 | CCFL_Man2 | i got my freebsd server on a 2U case in my rack now |
| 07:26.29 | tzafrir_home | infamy, what file system do you have in mind? |
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| 07:45.31 | loompek | exten => _0X.,1,Set(peerid=${IF($[${CALLERID(num)}=1]?230}) |
| 07:45.36 | loompek | does this seem okay? |
| 07:45.54 | loompek | if CALLERID(num) = 1 then set peerid=230 |
| 07:46.37 | ectospasm | loompek |
| 07:46.45 | yang | hey loompek man |
| 07:46.51 | loompek | ssup |
| 07:46.57 | ectospasm | when will callerid == 1? |
| 07:47.20 | loompek | if the call is from phone no.1 |
| 07:47.23 | loompek | anyways... |
| 07:47.39 | loompek | i have to set callerid(num) for every phone registerd... |
| 07:48.07 | ectospasm | that seems weird to me |
| 07:48.11 | loompek | http://rula.net/413 |
| 07:48.20 | loompek | this setup worked on asterisk 1.6 |
| 07:48.25 | loompek | but i've downgraded back to 1.4 |
| 07:48.37 | ectospasm | I don't know anything about 1.6 |
| 07:48.47 | ectospasm | and probably won't for a while |
| 07:49.02 | ectospasm | (Well, I know a little, but not enough to be useful or helpful) |
| 07:49.36 | loompek | too bad |
| 07:49.39 | loompek | yang any ideas? |
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| 07:52.39 | ectospasm | loompek: it looks like you're doing a lot of duplicate typing there, more prone to errors |
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| 07:56.19 | yang | loompek: sorry to early to think for me |
| 07:57.35 | *** part/#asterisk jklehm (n=john@67.53.187.205) |
| 08:03.10 | *** join/#asterisk jmhunter (n=jacob@72.14.86.68) |
| 08:03.10 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 08:03.21 | jmhunter | jbot |
| 08:03.28 | jmhunter | ~seen jbot |
| 08:03.30 | jbot | jbot is currently on #asterisk-doc (1d 4h 54m 4s) ##t42 (1d 4h 54m 4s) #how (1d 4h 54m 4s) #flyspray (1d 4h 54m 4s) #asterisk-dev (1d 4h 54m 4s) #asterisk (1d 4h 54m 4s) #byumug (1d 4h 54m 4s) #va (1d 4h 54m 4s) #orkut (1d 4h 54m 4s) #nslu2-linux (1d 4h 54m 4s) ##ducleague (1d 4h 54m 4s) #storm (1d 4h 54m 4s) #uphpu (1d 4h 54m 4s) #aegis (1d 4h ... |
| 08:03.35 | jmhunter | ~seen kram |
| 08:03.36 | jbot | kram <n=kram@cm56-214-79.liwest.at> was last seen on IRC in channel #debian, 208d 11h 51m 40s ago, saying: 'hey guys, my debian decided to show me parts of gnome and apt outout in half geran since a few weeks, can anyone tell me how to make it speak english eagain?'. |
| 08:04.12 | jmhunter | ~seen _bkw |
| 08:04.13 | jbot | _bkw <~zoa@p4.n-atpop03.stsn.com> was last seen on IRC in channel #asterisk, 1281d 14h 3m 10s ago, saying: 'lotsa packets a second ?'. |
| 08:04.28 | jmhunter | ~seen bkw_ |
| 08:04.29 | jbot | bkw_ <n=brian@ppp-70-128-106-102.dsl.tulsok.swbell.net> was last seen on IRC in channel #asterisk, 18d 17h 42m 7s ago, saying: 'thats ok.. would do it again in a heart beat'. |
| 08:04.59 | MrEIso | Can asterisk deal with incoming SIP NOTIFY's yet? |
| 08:05.03 | *** join/#asterisk shtoom (n=godson@121.246.167.147) |
| 08:05.11 | MrEIso | what documentation I can find suggests not. |
| 08:06.07 | ectospasm | MrEIso: what do you mean? |
| 08:06.36 | MrEIso | I have an upstream SIP provider that (annoyingly) does voicemail and sends me SIP NOTIFY's to tell me when I have voicemail |
| 08:07.05 | MrEIso | but asterisk as far as I can tell eats these notify's, and doesn't provide me with any way to notify downstream phones that I have voicemail waiting |
| 08:09.33 | ectospasm | do you know what the ring timeout for the upstream provider is? Just set the downstream phones to have a lower ring, and then when Comedian Mail answers the phone, the upstream provider thinks it's answered by something, and it never dumps to their voicemail system |
| 08:10.05 | ectospasm | at least, that's how I'd try to workaround the problem |
| 08:10.35 | MrEIso | yeah, done that. occasionally if the 'nets down or something I'll get a voicemail that gets stuck in my upstream |
| 08:11.18 | loompek | am i the onlyone that thinks that asterisk's if func is really really hard to read in case editor doesn't support matching bracket coloring/highlighting? |
| 08:12.08 | ectospasm | loompek: the same could be said for any computer language |
| 08:12.26 | ectospasm | Imagine trying to do Lisp without matching parentheses... |
| 08:12.53 | loompek | i don't do lisp... |
| 08:12.59 | ectospasm | have you ever seen it? |
| 08:13.09 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:13.10 | ectospasm | just look at som emacs code |
| 08:13.10 | loompek | exten => _0X.,1,Set(peerid=${IF($[${CALLERID(num)}=2]?231:IF($[${CALLERID(num)}=3]?232:230)}) |
| 08:13.18 | loompek | try figuring this out :D |
| 08:13.28 | ectospasm | pastes it into vim |
| 08:13.44 | ectospasm | or wait, is that against the rules? (-:< |
| 08:13.56 | loompek | :D |
| 08:15.01 | ectospasm | loompek, you need (at least) a closing parenthesis for the Set app |
| 08:15.11 | loompek | i prolly need more |
| 08:15.28 | loompek | installing notepad++ update... hopefully it will work okay in my vista |
| 08:16.15 | ectospasm | gak |
| 08:16.26 | ectospasm | Install Gvim |
| 08:16.30 | ectospasm | you'll be much happier |
| 08:16.52 | ectospasm | syntax highlighting, available for asterisk configs, etc. |
| 08:17.42 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 08:18.34 | ectospasm | loompek: always think of parentheses, brackets, and braces as a stack. Count from left to right (or right to left) and whenever you find a matching one, "pop" it off the stack, subtracting by one |
| 08:18.37 | loompek | not much of a 'vi' person... |
| 08:18.43 | loompek | using joe primarily |
| 08:18.56 | santoshr | I am having trouble dialing via the H323 channel. When i do a dial(h323/<no>@<ip>) i get the following mesage. |
| 08:18.58 | santoshr | <PROTECTED> |
| 08:19.04 | santoshr | the call disconnects |
| 08:19.26 | ectospasm | It might help to learn your way around vi, because on some systems it'll be all you got |
| 08:19.29 | santoshr | i am able to recieve calls using the chan_h323 channel. but dialing out is troublnig |
| 08:19.47 | ectospasm | I have no experience with H323 |
| 08:20.34 | ectospasm | have you tried any of the other h323 implementations? Last I heard ooh323 had the most H323 features built into it (but I could be WAAAY wrong here) |
| 08:24.48 | santoshr | i tried my way with sip and tht worked fine. |
| 08:25.25 | santoshr | ectospasm: agreed something could be wrong with chan_h323.so but then i wouldnt have been able to get a call into asterisk using that channel which i am able to |
| 08:25.33 | loompek | gakh... |
| 08:25.37 | loompek | i think i finally got it |
| 08:25.38 | loompek | exten => _0X.,1,Set(peerid=${IF($[ ${CALLERID(num)} = 2]?231:${IF($[ ${CALLERID(num)} = 3]?232:230)})}) |
| 08:26.25 | ectospasm | loompek, what happened to your '1' condition? |
| 08:26.46 | ectospasm | santoshr: not necessarily |
| 08:27.12 | ectospasm | santoshr: it's probably a configuration issue at that point |
| 08:27.43 | ectospasm | I don't even know how to debug it... is there a "h323 debug" CLI command?? |
| 08:29.44 | *** part/#asterisk MrEIso (i=dahoose2@60-234-141-149.bitstream.orcon.net.nz) |
| 08:30.28 | loompek | ectospasm i just set the default... |
| 08:30.48 | ectospasm | looks |
| 08:30.55 | loompek | if caller = 2, then set it to 231, if caller = 3, set it to 232, set 230 in every other case |
| 08:31.12 | ectospasm | I see, only 2 and 3 are the special cases |
| 08:31.24 | loompek | well there will be more |
| 08:31.29 | loompek | but no more than 20 :D |
| 08:31.41 | loompek | this is just a testing environment |
| 08:31.56 | loompek | or even better yet... |
| 08:32.13 | loompek | in the production environment i'll set users according to this dialplan |
| 08:32.29 | loompek | instead of [1] i'll set [230] directly |
| 08:47.38 | *** join/#asterisk jroysdon (n=jroysdon@jroysdon.broker.freenet6.net) |
| 08:54.17 | santoshr | ectospasm: ok found the issue. I had bound h323 to a specific port. SO when dialing out it was sending out to the ip at the port mentioned in the h323.conf. |
| 08:54.38 | santoshr | ectospasm: as u had said configuration issue it was. |
| 08:54.53 | ectospasm | santoshr: that's what I thought |
| 09:01.26 | yang | loompek: have you seen our local Users Group ? |
| 09:02.05 | loompek | yang check out the comments :D |
| 09:03.18 | yang | loompek: comment on the page or comment in IRC ? |
| 09:04.00 | yang | Becouse i see none |
| 09:06.34 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 09:06.58 | coppice | only the true believer sees |
| 09:07.08 | loompek | yang what's the url again? |
| 09:07.53 | Leland | morning scrubs ;-) |
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| 09:38.31 | mkillebrew | what's a good 802.11 SIP phone? |
| 09:41.32 | *** join/#asterisk stoffell (n=stoffell@d51A4DCAE.access.telenet.be) |
| 09:45.19 | coppice | there isn't a good one |
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| 10:02.09 | Unlockgod | Hey guys |
| 10:02.46 | Unlockgod | Anyone know why when i get an incoming call the IVR prompts break up reali bad |
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| 10:21.01 | loompek | mkillebrew nokia n95? :D |
| 10:23.07 | loompek | i've got bcm wilan800 but it preforms kinda sluggish |
| 10:24.01 | loompek | s/wilan/wlan/ |
| 10:25.19 | loompek | Executing [3@odjemalci:1] Dial("SIP/1-0822cb60", "SIP/3|60|rm") in new stack |
| 10:26.38 | loompek | this should like.. play both ringtone and music on hold.. right? |
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| 13:06.57 | markit | hi, I've updated 1.4.x from svn stable branch, and now compilation gives me the error: codec_ilbc.c:50:30: error: ilbc/iLBC_encode.h: No such file or directory Any clue? |
| 13:07.30 | markit | (debian 64 sid here) |
| 13:09.16 | lmadsen | markit: ilbc support was removed from the standard 1.4 distribution due to licensing reasons. I believe it got moved to -addons |
| 13:09.52 | markit | so I have to reconfigure asterisk, I guess? |
| 13:09.58 | lmadsen | yes |
| 13:10.06 | lmadsen | you might have to run make distclean first |
| 13:11.11 | markit | mm I see, is freeware, not Free Software, sigh |
| 13:11.21 | markit | thanks lmadsen |
| 13:12.04 | *** join/#asterisk javiolo (n=user@unaffiliated/javiolo) |
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| 13:24.26 | *** join/#asterisk DragoraN (n=dragoran@dial-249-242.orange.sk) |
| 13:24.39 | DragoraN | hi |
| 13:24.49 | DragoraN | is it possible to connect Avaya 9620 phone to asterisk? |
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| 13:40.47 | PepOSX | DragoraN, http://www.voip-info.org/wiki/view/Avaya |
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| 13:48.16 | boblutz | Is anyone here a Toronto FC supporter? |
| 13:49.42 | *** part/#asterisk boblutz (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
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| 13:54.31 | ZPertee | as long as I have this: Dial(SIP/1,t) and features.conf setup correctly I can start transferring calls right? |
| 13:55.31 | Katty | MORNING. |
| 13:55.34 | Katty | oh |
| 13:55.35 | Katty | morning |
| 13:56.08 | *** join/#asterisk AndyGraybeal_ (n=AndyGray@128.177.27.78) |
| 13:56.36 | ZPertee | top of the morning to you |
| 13:58.08 | ZPertee | is something wrong with this Dial(SIP/1,t) says that there is no "t" option. voip-info says there is |
| 13:58.20 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 13:59.07 | Katty | ZPertee: t is the timeout number, in seconds. |
| 13:59.26 | Katty | ZPertee: exten => 1,1,Dial(SIP/1,20) |
| 13:59.33 | Katty | ZPertee: then in 20 seconds it goes to |
| 13:59.41 | Katty | ZPertee: exten => 1,n,Something |
| 14:00.57 | ZPertee | Katty, this is what asterisk cli says "t - Allow the called party to transfer the calling party by sending the DTMF sequence defined in features.conf." |
| 14:01.20 | loompek | zirman try Dial(1/SIP,,t) |
| 14:01.30 | loompek | err |
| 14:01.32 | loompek | SIP/1 |
| 14:01.48 | Katty | yes, options come after timeout |
| 14:01.49 | loompek | Dial(SIP/1,,t) |
| 14:01.59 | ZPertee | loompek, ok if i skip options still put the command in though. that's what I was wondering |
| 14:02.26 | Katty | i think i have transfer options set in sip.conf for each user |
| 14:02.48 | Katty | maybe i'm thinking of something else tho |
| 14:04.54 | ZPertee | is there any way to just reload one sip user? I have an ata that won't hangup after I do a channelredirect |
| 14:06.02 | *** join/#asterisk shinao1 (n=shinao1@35-230.rv.ipnxtelecoms.com) |
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| 14:28.33 | *** join/#asterisk Unlockgod (n=dave@adsl-87-102-70-218.karoo.KCOM.COM) |
| 14:28.40 | Unlockgod | Hey everyone |
| 14:29.22 | Katty | hello |
| 14:30.29 | Unlockgod | Does anyone know why my voices on the IVR are breaking up |
| 14:30.45 | Unlockgod | voice calls work fine just the IVR section breaks |
| 14:39.39 | *** join/#asterisk bminish (n=bminish@brenbox.home.minish.org) |
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| 14:47.02 | keith4 | Unlockgod: load problem? |
| 14:47.25 | Unlockgod | Boots up fine |
| 14:47.31 | Unlockgod | just whe you ring the telephone number |
| 14:47.41 | Unlockgod | and it says "Welcome" in the default voice |
| 14:47.55 | Unlockgod | its very broken, i recored my own as well and that is very broke |
| 14:48.06 | Unlockgod | the ring ring sounds fine tho and the voice quiality is perfect in calls |
| 14:48.30 | Unlockgod | im using SIPGate as my provider |
| 14:48.46 | keith4 | could be lots of things |
| 14:49.06 | keith4 | if you're convinced the server isn't bogged down, cpu-wise, IO-wise, etc., then it could be network related |
| 14:49.18 | keith4 | like...jitter |
| 14:49.26 | Unlockgod | i cant understand why its just the IVR voice |
| 14:49.27 | Unlockgod | and nothing else |
| 14:50.38 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
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| 15:08.39 | ZPertee | what happens if someone trys to dial out on a zap channel that already has an incoming call? |
| 15:09.55 | ZPertee | the others on the phone just here someone trying to dial, same as when my little sisiter messes with the phone when I am on it? |
| 15:13.09 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
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| 15:22.23 | ZPertee | can someone tell me if this code http://pastebin.ca/962155 looks valid or not. I can't test it right now |
| 15:28.03 | DragoraN | ZPertee: from inside or from outside? |
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| 15:32.10 | ZPertee | DragoraN, the idea is that from the inside I dial 9 and then the number. it lets me listen for five seconds to make sure there isn't already a call on that zapchannel and then proceeds to make the call. if someone is already on the zap channel I can just hangup |
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| 15:47.37 | drmessano | It should already tell you the line is busy |
| 16:02.10 | ZPertee | drmessano, ok so thats buit in then? If Itry to dial out on an a line in use I'll get a busy tone or what? |
| 16:03.34 | *** join/#asterisk ManxPower (n=manxpowe@120.sub-75-201-90.myvzw.com) |
| 16:03.54 | drmessano | Have you TRIED it? |
| 16:04.06 | *** join/#asterisk weazahl (n=jeremy@12.53.40.34) |
| 16:07.50 | *** join/#asterisk ceegee (n=christia@p57BD77EB.dip.t-dialin.net) |
| 16:07.53 | ceegee | hello |
| 16:09.07 | ceegee | I am a liite confused an required packages for building asterisk. do I need to install libpri and zaptel in every case, also when I dont have pri and digium cards? |
| 16:10.58 | ZPertee | ceegee, you don't have cards digium or otherwise? |
| 16:11.20 | ceegee | ZPertee: we arfe going to buy a beronet bn8s0 card |
| 16:11.45 | ceegee | so I only need asterisk and misdn stuff, right? |
| 16:13.33 | ZPertee | ceegee, not sure, I only use digium stuff |
| 16:14.07 | ZPertee | ceegee, as a guess I would agree with you though, sounds logical |
| 16:14.53 | ceegee | I will try it without any special hardware on a virtual machine |
| 16:15.15 | ZPertee | ceegee, ok sounds good. |
| 16:16.44 | *** join/#asterisk DonAlex (n=DonAlex@mirafiori.demon.co.uk) |
| 16:16.54 | DonAlex | Hey all.. :) |
| 16:17.26 | DonAlex | Ok just a quick question this time.. but one that I feel is kinda fundamental to my understanding of asterisk. |
| 16:18.11 | DonAlex | In what circumstances would asterisk only read context information from extensions.ael and completely ignore anything in extentions.conf ? |
| 16:18.35 | madduck | DonAlex: when extensions.conf does not exist |
| 16:18.57 | DonAlex | madduck: And when it does? |
| 16:19.11 | DonAlex | madduck: What are the permissions on it supposed to be ? |
| 16:19.37 | madduck | that depends on your need. If * is running as a non-root user, that user needs read access. |
| 16:19.55 | madduck | DonAlex: if it exists and is empty, then it's also basically ignored |
| 16:20.03 | madduck | if it's non-empty, it'll be read and used |
| 16:20.19 | DonAlex | madduck: hmm but both extensions.ael and extensions.conf have the same permissions? |
| 16:20.31 | madduck | good |
| 16:21.06 | DonAlex | madduck: is it trying to tell me that my extensions.conf is totally invalid? |
| 16:21.35 | DonAlex | madduck: I just don't quite buy that as it was copied from a working installation? |
| 16:21.47 | madduck | i have no idea what it's trying to tell you |
| 16:22.47 | DonAlex | madduck: Well anyway to debug what files asterisk is parsing as it starts up? debug =5 does not seem to give that |
| 16:22.49 | weazahl | is this a problem with signaling? "== Primary D-Channel on span 2 down" |
| 16:24.02 | madduck | no idea |
| 16:24.36 | DonAlex | madduck: *sighs* Fairy nuff.. I am sure I am missing something fundamental though.. |
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| 16:42.28 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:43.45 | ManxPower | weazahl: that means your PRI went down |
| 16:44.09 | ManxPower | That is frequently caused by having too different of the version number between Asterisk/Zaptel/Libpri |
| 16:44.27 | ManxPower | If that's not the problem then you either have a sync source problem, a cable problem, or a telco problem |
| 16:44.41 | lesouvage | Maybe off topic, but I need to join (not to mix) three .wav files. Does anybody knows how to this with a line command . I found cat file1.wav file2 > new_file.wav but, although the files are joined only the first one is played. |
| 16:45.04 | ManxPower | lesouvage: sox |
| 16:46.06 | DonAlex | Something really really must be so bloody obvious I am missing it.. But I cannot see it loading or parsing extensions.conf ANYWHERE despite running asterisk -vvvvvvvvvgc |
| 16:46.29 | lesouvage | ManxPower: I tried several ways with sox but I seem to miss something. Could you please give me a hint. |
| 16:46.50 | DonAlex | does loading extensions.ael override reading extensions.conf or something? |
| 16:47.10 | weazahl | ManxPower: it is signalling im sure. i am having a hard time making the sangoma speak avaya |
| 16:47.17 | ManxPower | lesouvage: no. I would have to read the docs for sox |
| 16:47.40 | ManxPower | weazahl: Either the line is PRI or it's not. |
| 16:48.16 | DonAlex | *sobs* This is driving me nuts.. :( |
| 16:48.39 | tzafrir_home | weazahl, is layer 1 working? (if not: you have red alarm) |
| 16:49.12 | DonAlex | tzafrir: Hey Tza? Maybe you can help me from going loopy? |
| 16:49.28 | weazahl | i have been able to get green condition almost all the time but the avaya and * never speak |
| 16:49.44 | outtolunc | parsing of extensions.conf (then extensions.ael) is top down, meaning it will parse, load inorder into dialplan.. |
| 16:49.52 | tzafrir_home | DonAlex, not exactly sure of the interaction between the two |
| 16:50.15 | tzafrir_home | Do you have the same context defined in both? |
| 16:50.36 | DonAlex | outtolunc: Yeah... well I manual unloaded the ael module and now it is not registering a dial plan at all, save for the parked calls in the features extension |
| 16:51.28 | DonAlex | tzafrir: I just cannot figure out why it is not even attempting to parse extensions.conf? I mean it is like it does not exist only I know it is a good config cause it came of the XR1000 ;) |
| 16:51.48 | outtolunc | if you are unaware of where your contexts for a particular item is at, you are obviously using some type of gui for management |
| 16:53.09 | DonAlex | outtolunc: Partly sure.. Which is why I am not running asterisk in debug mode to try and figure out what is missing. I mean it is not even attempting to parse the file let alone kick up any missing contexts.. Surely that is a bigger problem? |
| 16:53.30 | DonAlex | outtolunc: Sorry I AM running in debug mode.. |
| 16:53.51 | *** join/#asterisk xenonex (n=xenonex@92.47.14.6) |
| 16:54.14 | outtolunc | an example would be with freepbx, you do not put your mods in extensions.conf you put them in extensions_additional.conf which is *included* in extensions.conf... but that is all mute as we do not support gui's here |
| 16:55.15 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:55.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:57.31 | DonAlex | outtolunc: Well tza will know the system I am using.. but basically it is based around the guitools from Digium.. And as far as I can see they are intact else the gui would not work. But even Manually entering contexts it seems to just not even try to read the file.. |
| 16:58.19 | outtolunc | if it is based on the digium gui, then no it does not use extensions.conf (iirc) as it was 100% (as close as possible) AEL only |
| 16:58.43 | outtolunc | so what you are experiencing is correct |
| 16:58.50 | ZPertee | is there any way to make the voicemail have less options? seems like it asks too many press this for that |
| 16:59.10 | DonAlex | outtolunc: If that were so then why is it ADDING the contexts to the .conf file? |
| 16:59.33 | outtolunc | sounds like something is misconfigured <G> |
| 16:59.54 | DonAlex | outtolunc: But I am right it is not even looking at the extensions.conf I just PUT a great be ERROR THIS IS AND ERROR at the top of it and it did not mention it |
| 16:59.55 | outtolunc | like that you unloaded ael, so what is asterisk to do now... PUNT |
| 17:00.24 | DonAlex | outtolunc: Actually I out ECHO THIS IS AN ERROR.. my typing sucks it seems |
| 17:00.27 | DonAlex | outtolunc: :) |
| 17:00.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:02.00 | outtolunc | well considering what has/hasnot/isnot happening, i would definitely contact digium on monday (unless they are open now) to have someone take a peek at your setup |
| 17:02.02 | ManxPower | DonAlex: and that is why we don't support GUIs here, go to the correct channel |
| 17:02.49 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:03.05 | outtolunc | mmmm's coffee |
| 17:03.13 | DonAlex | ManxPower: Ok.. so say I forget about the GUI.. can someone still explain to me why editing extensions.conf is ignored even when I am NOT loading the ael module? |
| 17:03.30 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 17:03.34 | DonAlex | ManxPower: Which is what is happening currently? |
| 17:05.49 | ManxPower | DonAlex: We can't help you with ANY config files, as the GUIs totally screw them up. |
| 17:05.52 | ManxPower | ~freebpx |
| 17:05.55 | ManxPower | this is why |
| 17:06.06 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 17:06.48 | ManxPower | DonAlex: if you want to forget about the gui then delete all the files in /etc/asterisk and build them from scratch. The DIFFERENCE between Asterisk and GUIS for Asterisk ARE the config files. |
| 17:06.59 | DonAlex | ManxPower: Yeah but this is crazy! Cause I just copied the debian dist.pkg extension.conf and that is being ignored too.. |
| 17:07.54 | DonAlex | ManxPower: All I really need to know is WHY is it not reading the default file at all?! I mean where the hell does it configure asterisk to do that?! |
| 17:08.05 | DonAlex | ManxPower: Never even thought it was possible.. let alone configurable |
| 17:08.12 | russellb | if pbx_config is loaded, it will read it |
| 17:08.28 | DonAlex | russellb: pbx_config? |
| 17:08.33 | russellb | that is the module name |
| 17:08.33 | DonAlex | hmmm |
| 17:08.53 | DonAlex | russellb: Methinks you have hit the nail on the head my man.. |
| 17:09.08 | weazahl | PRI is robbed bit or CC? |
| 17:09.28 | russellb | CC |
| 17:09.32 | weazahl | thanks |
| 17:10.23 | DonAlex | HOOPLA!!! |
| 17:10.50 | DonAlex | russellb: Wondrful Russ.. I just KNEW it was such a obvious thing.. |
| 17:11.01 | russellb | :) |
| 17:12.04 | DonAlex | Now can some genius tell me what the hell is pbx_config doing in a module?! I mean abstraction is one thing but oh.. never mind.. .Live and learn huh... |
| 17:12.18 | DonAlex | Thanks for all your assistance peeps.. Catch ya round. |
| 17:13.09 | russellb | blinks |
| 17:14.22 | *** join/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 17:14.34 | outtolunc | yeah i seemed to remember asterisknow was ael only |
| 17:14.40 | russellb | it is not |
| 17:14.43 | russellb | it does not use AEL at all |
| 17:15.41 | outtolunc | hmm, then i have cross-gui-itis ... need to remember which one was |
| 17:18.36 | boblutz | still unsure of what the gui does |
| 17:19.14 | boblutz | ~root |
| 17:19.14 | jbot | [root] not a Good Thing to use when using IRC. Please use a different account. |
| 17:19.46 | boblutz | Would anyone happen to recall what chapter in ~thebook refers to running Asterisk as non-root? |
| 17:22.13 | ManxPower | Good thing DonAlex stayed and pestered us instead of going to the right channel or he may never have gotten an answer! |
| 17:22.44 | russellb | asking how extensions.conf gets read is a pretty straight forward asterisk question |
| 17:23.08 | ManxPower | russellb: It would have been loaded automatically if he had not been using weird config files. |
| 17:23.41 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 17:28.01 | ManxPower | I think most of us get sick and tires of trying to help someone only to find things like all configs in users.conf or a million AGI scripts in extensions.conf and/or chan_zap not being built in an Asterisk package |
| 17:28.35 | boblutz | ManxPower: Or trying to trouble shoot phantom calls on a tdm410p with an rj-14 line ? |
| 17:28.44 | boblutz | smirks |
| 17:28.53 | ManxPower | boblutz: That problem was mainly the user refusing to take the advice. |
| 17:29.16 | boblutz | you are right |
| 17:30.34 | ManxPower | did the user ever solve the problem? |
| 17:30.51 | boblutz | 2+ hours on the phone with digium and we came to the conclusion its a bad pci card |
| 17:31.06 | boblutz | placing the x100m's on the old tdm400 would not reproduce the issue |
| 17:32.49 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:35.22 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 17:36.54 | eric2 | what's the best way to configure the linksys 2101 for fail over if you have 2 identical asterisk servers running? |
| 17:38.24 | Bourrelle | Hello, ive read on the internet about this but no luck :( |
| 17:38.36 | Bourrelle | when asterisk tries to send me video |
| 17:39.00 | Bourrelle | it says ICMP Protocol Unreachable |
| 17:39.10 | Bourrelle | Any idea on how to open a port under linux |
| 17:39.21 | boblutz | ICMP is the port you ping on, right? |
| 17:40.00 | ManxPower | ICMP has no ports |
| 17:40.22 | boblutz | err ICMP protocol is essentially "pinging" ? |
| 17:40.33 | boblutz | `ping` uses the ICMP protocol |
| 17:40.35 | ManxPower | ICMP is what ping uses, as well as being generated when you try to send a packet to a host, network, or port that has nothing listening on it. |
| 17:41.27 | ManxPower | Ping uses ICMP type 8 |
| 17:41.47 | Bourrelle | it says it says ICMP Protocol Unreachable (PORT UNREACHABLE) |
| 17:41.55 | boblutz | firewall issue perhaps? |
| 17:42.04 | ManxPower | Bourrelle: sounds like nothing is listening on something somewhere, but we really can't help much until we get more information |
| 17:42.32 | ManxPower | Bourrelle: is there anything listening on port 5060? |
| 17:42.38 | ManxPower | UDP 5060, of course |
| 17:42.51 | Bourrelle | of course |
| 17:43.01 | Bourrelle | audio is working great ! |
| 17:43.04 | ManxPower | Bourrelle: and you have no NAT or firewall involved? |
| 17:43.19 | Bourrelle | but video, when asterisk tries to forward those H.263 packet |
| 17:43.28 | Bourrelle | it says port unreachable |
| 17:43.29 | Bourrelle | no firewall |
| 17:43.32 | Bourrelle | no Nat |
| 17:43.37 | ManxPower | and what port is it trying to reach? |
| 17:43.44 | ManxPower | and what client are you using? |
| 17:43.52 | Bourrelle | Its a ramdon port each time |
| 17:43.57 | Bourrelle | just like audio |
| 17:44.12 | ManxPower | those are not random ports |
| 17:44.31 | ManxPower | Now, please answer my question, rather than arguing that the question is not relevant |
| 17:45.06 | ManxPower | If some information I request is not easy to get tell me and I'll see if there is another way to get the info. |
| 17:45.20 | Bourrelle | ok |
| 17:45.30 | ManxPower | ManxPower: and what client are you using? |
| 17:46.23 | Bourrelle | sflPhone |
| 17:47.36 | ManxPower | Perhaps this page will be helpful (it says video support is not done): http://www.sflphone.org/index.php?mod=features |
| 17:49.05 | ManxPower | http://blog.tmcnet.com/blog/tom-keating/voip/sflphone-open-source-sip.asp notice one of the planned features is video support |
| 17:49.18 | Bourrelle | thanks alot |
| 17:49.21 | Bourrelle | ill go read all this |
| 17:49.37 | ManxPower | This Google Proxy says you're welcome |
| 17:52.41 | ManxPower | supposedly XTen EyeBeam works, but I have not used it personally |
| 17:54.26 | coppice | is sflphone still going? |
| 18:02.42 | weazahl | i have GREEN on my PRI on * side and RED on the avaya!?! |
| 18:04.56 | weazahl | shouldnt i have more packets than this? http://pastebin.com/m56d8bff2 |
| 18:05.44 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 18:07.11 | Katty | hello |
| 18:07.18 | *** join/#asterisk coolblade (n=gregoryr@166-70-57-81.ip.xmission.com) |
| 18:07.44 | coolblade | do you have to do anything after altering the sip.conf file? |
| 18:08.13 | boblutz | go to work on monday probably |
| 18:09.11 | Katty | do do do |
| 18:09.13 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 18:09.20 | *** part/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 18:11.42 | boblutz | toronto fc vs crew in 2 hours - i got my ticket in hand |
| 18:12.30 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 18:14.31 | Katty | i should do laundry |
| 18:15.26 | plik | katty: "please enter the spin-speed you require, forrlowd by the hash key" |
| 18:15.35 | Katty | meh |
| 18:15.50 | plik | erk s/forrlowd/followed |
| 18:16.20 | plik | goes back to sleep |
| 18:16.37 | boblutz | places plik's hand in a bucket of warm water |
| 18:16.40 | boblutz | LOL watch this |
| 18:17.28 | plik | accidentally flips the bucket of water over boblutz's asterisk boxen |
| 18:17.35 | plik | oops |
| 18:17.40 | boblutz | no!!! im melting! |
| 18:17.48 | plik | & |
| 18:18.36 | Katty | someone motivate me to do laundry |
| 18:18.45 | boblutz | u smell bad |
| 18:18.52 | Katty | scowls |
| 18:18.57 | Katty | i smell just fine, thank you |
| 18:19.15 | outtolunc | if you use it, it will break, and therefore get replaced with a newer more automated version <G> |
| 18:19.30 | Katty | if i do laundry... |
| 18:19.34 | Katty | eventjually it will automate itself? |
| 18:19.38 | Katty | <:) |
| 18:19.43 | Katty | boingboing, runs off to do laundry |
| 18:19.46 | boblutz | o.0 |
| 18:19.47 | outtolunc | sure, teach the kids to do it |
| 18:20.57 | Katty | kids? |
| 18:21.01 | Katty | are you insane? |
| 18:21.09 | outtolunc | yes |
| 18:21.18 | Katty | how could anyone wish that much pain and frustration upon themselves. |
| 18:21.20 | *** join/#asterisk kamitodo (n=kmt@pool-141-156-248-50.res.east.verizon.net) |
| 18:21.22 | Katty | laundry is bad enough. |
| 18:21.28 | boblutz | Katty: Ask your parents? |
| 18:21.42 | outtolunc | i believe anything (including teaching kids to wash their own clothes) is possible <G> |
| 18:22.18 | outtolunc | notes if using the proper voltage |
| 18:22.21 | Katty | boblutz: my mom's just insane.l |
| 18:22.31 | Katty | outtolunc: my only 'kids' are 4 furry ferrets |
| 18:22.43 | coolblade | hey guys I am trying to set up a remote sip, after I alter the sip.conf is there anyway I can troubleshoot to see if there are errors like login errors? |
| 18:29.56 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:30.47 | *** join/#asterisk jcaceres (n=asd@182-44-89.adsl.terra.cl) |
| 18:30.50 | jcaceres | helllo |
| 18:30.55 | jcaceres | hello* |
| 18:31.07 | boblutz | jcaceres: hay |
| 18:33.00 | jcaceres | i have some kind of error with a tdm404, (4fxo), when i send a call to it, the sounds of the tones that the card sends are noisy |
| 18:33.30 | jcaceres | an i assume that that's the reason why the calls are not been done |
| 18:33.39 | boblutz | jcaceres: Out of curiosity, is the pci card a tdm410 or tdm400 |
| 18:33.40 | jcaceres | any idea in how to solve this |
| 18:33.57 | jcaceres | tdm400 |
| 18:35.02 | jcaceres | properly i think its tdm404, because it's tdm400 with 4 fxo modules |
| 18:35.20 | jcaceres | any idea? |
| 18:35.36 | coolblade | is sip show not a command to run in asterisk? I see it on a few webpages but I get: No such command 'sip show' (type 'help' for help) |
| 18:36.24 | coolblade | jcaceres: fxo modules are goint to the pstn right? |
| 18:36.24 | jcaceres | and somthing i forgot to mention is that i have 3 cards of the same kink in the same server |
| 18:36.53 | jcaceres | no they are goint to a cellphone adapter |
| 18:37.40 | jcaceres | that sends me the tone, i have tested it with a normal analog phone and it works ok |
| 18:38.07 | coolblade | how are you connecting the analog phone? |
| 18:38.14 | jcaceres | but when the card sen the tones ther sound noisy |
| 18:38.52 | jcaceres | coolblade: directly to the cellphoneadapter |
| 18:39.27 | jcaceres | just a 2 wires cable, and rj11 connectors |
| 18:40.49 | boblutz | rj-14? |
| 18:45.23 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:45.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:48.35 | coolblade | anyone know what this means: [Mar 30 02:42:13] NOTICE[306] chan_sip.c: Unable to load config sip.conf |
| 18:48.35 | boblutz | yea i hear you |
| 18:48.51 | boblutz | alright well just give me a call when you get it worked out |
| 18:48.53 | *** part/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 18:50.56 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 18:51.07 | ThatKidKel | anyone got a recommendation for a T.38 termination provider? |
| 18:53.43 | *** join/#asterisk RoyK (n=roy@ti211110a081-5006.bb.online.no) |
| 18:54.56 | *** join/#asterisk nvicf (n=Vic@201.250.161.203) |
| 18:55.42 | jcaceres | i have a newbie question, how can i take the fxo line to send the tones directly during the conversation |
| 18:55.52 | jcaceres | not like a disa |
| 18:56.41 | jcaceres | just taking the line to make a direct dial on it |
| 18:56.43 | jcaceres | any idea? |
| 19:02.01 | jroysdon | uhm, pause? |
| 19:02.36 | jroysdon | oh, n/m, wrong channel.. I thought I was in #cisco ;-p |
| 19:02.57 | jcaceres | ¬¬ |
| 19:03.17 | jroysdon | On a Cisco router, I'd pick up the trunk, dial my number, and then add ,,1234 (the commas make it pause 2 seconds) |
| 19:03.30 | jroysdon | Works great to talk to gates, paging systems, and the like. |
| 19:11.08 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 19:11.08 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:11.16 | *** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 19:14.21 | jcaceres | i am doing some test with fxotune, but i couldn't make it work |
| 19:14.58 | jcaceres | the channel still sending noisy dtmf |
| 19:18.16 | jcaceres | i think i have an error with fxo tunes, because when i do it i cann not find the channel |
| 19:21.42 | jcaceres | ? |
| 19:21.51 | jcaceres | <PROTECTED> |
| 19:22.10 | jcaceres | [trixbox1.localdomain ~]# ls /dev/zap/ |
| 19:22.10 | jcaceres | 1 10 11 12 2 3 4 5 6 7 8 9 channel ctl pseudo timer transcode |
| 19:22.20 | coolblade | google is so annoying today |
| 19:22.49 | jcaceres | why do you say that? |
| 19:22.59 | coolblade | the homepage is all black |
| 19:23.05 | coolblade | for lights out day |
| 19:24.11 | *** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com) |
| 19:26.48 | jcaceres | not arround here |
| 19:27.25 | jcaceres | well? any idea why fxotunes doesnot find the channels even when they are listed? |
| 19:30.54 | tzafrir_home | anybody still have problems with zaptel on kernel 2.4? |
| 19:31.16 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 19:31.57 | tzafrir_home | jcaceres, listed where? |
| 19:32.15 | coolblade | jcaceres, sorry i am a newb |
| 19:32.26 | kombi | someone in here yesterday said that app_jack is in 1.6 which is technically right, only the option in menuselect is XXX. What's with the three Xs? |
| 19:32.41 | tzafrir_home | jcaceres, it could be that they are in use |
| 19:32.55 | tzafrir_home | cat /proc/zaptel/* and see if the channel is '(In use)' |
| 19:33.11 | tzafrir_home | (which would typically be by asterisk) |
| 19:33.29 | tzafrir_home | kombi, porn? |
| 19:33.46 | kombi | tzafrir_home: ??? |
| 19:33.55 | tzafrir_home | NM |
| 19:33.59 | kombi | oh, I get it.., no...;) |
| 19:34.02 | Cresl1n | jcacres: what happens if you cat /dev/zap/1 |
| 19:34.14 | Cresl1n | sorry, jcaceres |
| 19:34.39 | coolblade | quick question, my sip provider is telling me my user is my DIDs, which I have 3, any idea how I would format this, would it be 999-999-9999,999-999-9998,999-999-9997 (with commas) or with some other seperator |
| 19:34.56 | kombi | tzafrir_home: no problem, but what does one do enable an application that is marked like that in menuselect? |
| 19:35.03 | kombi | do to.. |
| 19:35.48 | tzafrir_home | I figure it depends on some components you don't have (disabled? not detected by configure?) |
| 19:36.04 | ManxPower | coolblade: your question makes no sense unless you tell you WHAT you are formatting it for. |
| 19:36.16 | kombi | tzafrir_home: that might be absolutely right... thanks! |
| 19:36.24 | coolblade | manxpower, sorry trying to configure my sip.conf |
| 19:36.24 | jcaceres | oka |
| 19:36.31 | ManxPower | coolblade: In a Dial() statement, registration, etc |
| 19:36.37 | jcaceres | let me see @Cresl1n |
| 19:36.54 | ManxPower | coolblade: you have three accounts, you need three register => lines and three [mydidhere] sections |
| 19:37.13 | Cresl1n | jcacres: did you email me the other day about this? |
| 19:37.16 | ManxPower | I would assume that you don't want anything except numbers, but that is TOTALLY up to how your provider set up your account name |
| 19:37.18 | Cresl1n | I'm Matthew Fredrickson at Digium |
| 19:37.20 | jcaceres | [trixbox1.localdomain ~]# cat /dev/zap/1 |
| 19:37.21 | jcaceres | cat: /dev/zap/1: No existe el dispositivo o la dirección |
| 19:37.25 | Cresl1n | yeah |
| 19:37.37 | Cresl1n | your channel looks like it's not configured in zaptel.conf or something of that nature |
| 19:37.38 | coolblade | manxpower, ok that is what i thought... I tried to just register one and I am getting: chan_sip.c: -- Registration for '801-386-7600@laext.sbc.primusvoip.com' timed out, trying again |
| 19:37.45 | ManxPower | jcaceres: you don't have the drivers loaded, you did not run ztcfg or your config is wrong |
| 19:37.53 | ManxPower | coolblade: what did I tell you about non-numbers? |
| 19:37.55 | Cresl1n | that too :-) |
| 19:38.11 | ManxPower | coolblade: usually a timeout means you have the wrong host you are connecting to. |
| 19:38.39 | ManxPower | coolblade: I didn't know Primus allowed non-primus provided hardware/software |
| 19:38.42 | coolblade | manxpower, thats what I thought to, but i pinged it and it pings fine |
| 19:38.54 | ManxPower | coolblade: pinging and not listening on a port are two different things. |
| 19:38.56 | coolblade | manxpower, they allow asterisk now |
| 19:39.08 | coolblade | manxpower, great point |
| 19:39.14 | ManxPower | coolblade: good. Let me know what happens when you remove all the extra chars |
| 19:40.27 | ManxPower | also, you normally register as register => username:password@hostname |
| 19:49.21 | coolblade | could it be a router or firewall issue? |
| 19:51.42 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 20:13.08 | *** join/#asterisk rafiks (n=rafiks@c-68-56-151-58.hsd1.fl.comcast.net) |
| 20:13.31 | rafiks | hi ..just wanna clear up something.. |
| 20:14.16 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 20:14.19 | rafiks | should i put my fqdn in externip rather than externhost |
| 20:15.06 | rafiks | whats the difference between the two? |
| 20:15.22 | rickross | anyone know how to build asterisk-addons properly for valgrind? The options in "make menuselect" for asterisk are not available for asterisk-addons |
| 20:16.14 | rickross | apparently someone else has encountered same problem, but I don't see a resolution - http://bugs.digium.com/view.php?id=11291 |
| 20:17.44 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 20:22.05 | coolblade | now I am getting: chan_sip.c: Got 404 Not found on SIP register to service IS78202466@asnjla3.bw.iprimus.net, giving up. Is that better than the timeout? |
| 20:23.18 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@20150228086.user.veloxzone.com.br) |
| 20:28.26 | rickross | anyone? |
| 20:30.36 | *** join/#asterisk ZPertee (n=ZPertee@cpe-65-189-208-20.neo.res.rr.com) |
| 20:32.55 | *** part/#asterisk jroysdon (n=jroysdon@jroysdon.broker.freenet6.net) |
| 20:36.05 | *** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com) |
| 20:37.41 | *** join/#asterisk bmg505 (n=leon@196-209-76-145-tbnb-esr-2.dynamic.isadsl.co.za) |
| 20:41.41 | coolblade | ManxPower, finally got it done, primus routed it to my asterisk box no problem |
| 20:51.10 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 20:52.58 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 20:58.53 | *** join/#asterisk R-MAN (n=no@78.149.164.4) |
| 20:58.56 | R-MAN | hey guys |
| 20:59.07 | ZPertee | what's up |
| 20:59.32 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 20:59.42 | R-MAN | im having a really hard time in getting sipgate outbound calls working anyone care to help? |
| 21:00.06 | siya | Anyone know how I can refer to cid_number in my dialplan? |
| 21:00.44 | siya | R-MAN: I only have sipgate inbound |
| 21:01.33 | tzafrir_home | what's cid_number? from where? |
| 21:01.35 | siya | R-MAN: Does inbound work? |
| 21:01.39 | R-MAN | I got incoming to work yes |
| 21:01.47 | tzafrir_home | ${CALLERID(num)}? |
| 21:01.56 | R-MAN | I dont get dtmf but thats another issue lol |
| 21:02.08 | siya | tzafrir_home: it's a custom callerid(num) for an extension |
| 21:02.21 | R-MAN | when I dial out I just get all circuits are busy |
| 21:02.41 | siya | tzafrir_home: erm... |
| 21:03.02 | R-MAN | ive checked alot of configurations online but not change |
| 21:03.10 | tzafrir_home | R-MAN, Please give a CLI trace from your call |
| 21:03.11 | hesco | I have an ADA from vonage. Is there any easy way to make it serve my need for bridging from an iax2 outbound trunk to an analogue phone in my office, w/o a vonage account? |
| 21:03.40 | R-MAN | hesco buy a SPA 3102 |
| 21:03.51 | R-MAN | pstn/pots to voip adaptor |
| 21:03.53 | R-MAN | gateway |
| 21:04.09 | R-MAN | brb time for some food |
| 21:05.09 | siya | tzafrir_home: actually you're completely right... |
| 21:05.33 | siya | goes to wash his eyes |
| 21:08.51 | siya | tzafrir_home: got it! |
| 21:09.09 | tzafrir_home | What is it? |
| 21:09.17 | siya | exten = s,n,set(CALLERID(all)=${IF($["${LEN(${CALLERID(num)})}" < "6"]?${CALLERID(all)}:${ARG2})}) |
| 21:09.26 | siya | the < was a > |
| 21:09.59 | siya | so the extension callerid(num) set was over written rather than honored |
| 21:12.34 | siya | I was under the impression cid_number didn't get set as callerid(num) => doh! |
| 21:15.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:20.12 | siya | keeps figuring out what he's doing |
| 21:21.07 | siya | I've just noticed that when I send CALLERID(all) instead of CALLERID(num) when dialling out UK networks do present my CLID |
| 21:21.23 | siya | but the line of code I posted is far from flawless... |
| 21:21.55 | defsdoor | siya ? |
| 21:22.31 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 21:22.39 | siya | defsdoor: yes? |
| 21:29.46 | drmessano | Twitter is now doubling as an IRC early warning system too |
| 21:29.48 | drmessano | <PROTECTED> |
| 21:30.08 | drmessano | Expect him at any time now |
| 21:34.51 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 21:38.23 | hesco | is there a way from the console to initiate an extension's dial plan? |
| 21:39.36 | hesco | I'm now experimenting with MeetMe for the first time and hoping to have an extension initiate a phone call between two outbound phone numbers, from my issuing a command to the console. Is that possible? |
| 21:48.00 | ectospasm | hesco: possible probably with a call file |
| 21:50.42 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 21:51.39 | hesco | ectospasm: thanks. Anyway from an *CLI> prompt directly? |
| 21:51.45 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:52.09 | ectospasm | hesco: create the call file outside of asterisk. then in the asterisk CLI: |
| 21:52.24 | ectospasm | <PROTECTED> |
| 21:52.51 | ectospasm | actually creating the call file is left as an exercise for the reader (-; |
| 21:53.20 | hesco | And the .call file would simply dial the entension which uses the MeetMe() conference, I suppose. |
| 21:53.28 | ectospasm | (see doc/callfiles.txt) |
| 21:53.47 | ectospasm | you can do pretty much anything in the dialplan with call files, afaik |
| 21:53.48 | hesco | I'll try that exercise now. I've used .call files to call a single number, but not multiple numbers. |
| 21:54.13 | ectospasm | well, if you can do it in dialplan, just do it there, and have the call file execute that part of the dialplan |
| 21:54.45 | hesco | your reference to doc/callfiles.txt brings me back to an unanswered question from yesterday: |
| 21:55.21 | hesco | I have asterisk 1.2.13 installed from the debian repository, but I'd like to try building from upstream source. |
| 21:55.58 | hesco | Can I somehow do that without tearing out or breaking a working installation, while I experiment with the newer version? |
| 21:56.10 | ectospasm | not easy |
| 21:56.37 | ectospasm | I highly suggest you create a development machine, copy the existing configs to that, and massage it into a working state. |
| 21:56.59 | ectospasm | It's doable, but if you're talking about a production system, it'd be stupid to do a blind in-place upgrade |
| 21:58.03 | supjigator | Do the google cnam scrapers work well? |
| 21:58.26 | adorah | Did anyone encounter a problem updating zaptel driver to current with TE120B? |
| 21:58.49 | hesco | This box is used only for asterisk. But I have no incoming DID's defined yet. I'm only using it for outbound calls and its used to run a few jobs each week, but is not in constant use. So I could shut down the non-working version and restart the working version when the need arose, until I tweaked out the 1.4 or the 1.6 installation. |
| 22:00.06 | ectospasm | adorah: what is the problem you're having? |
| 22:00.29 | ectospasm | hesco: messy |
| 22:00.38 | adorah | <ectospasm>After upgrade it doesn't recognize the hardware |
| 22:00.51 | ectospasm | what do you mean, it's not in lspci? |
| 22:01.01 | adorah | it is in lspci |
| 22:01.24 | adorah | but zaptel doesn't rcognize it |
| 22:01.41 | ectospasm | does dmesg show it when you load zaptel? What do you mean,"does not recognize it?" |
| 22:01.43 | tzafrir_home | adorah, is the relevant module loaded? |
| 22:01.54 | tzafrir_home | lsmod | grep ^zaptel |
| 22:01.59 | adorah | yes |
| 22:02.16 | adorah | oh I didn't try that comand |
| 22:02.21 | tzafrir_home | What do you have in /proc/zaptel/* |
| 22:02.21 | adorah | <PROTECTED> |
| 22:02.56 | tzafrir_home | adorah, I start diagnozing zaptel problems with: zaptel_hardware and lszaptel |
| 22:03.19 | adorah | I will do tomorrow a fresh install it is not a production server but I heard report regarding problems with othe Digium hardware with latest upgrades |
| 22:03.21 | hesco | ectospasm: yes, messy perhaps, but more economically feasible than buying new hardware at this moment, I'm afraid. |
| 22:03.38 | hesco | So does messy mean possible? |
| 22:03.39 | ectospasm | hesco: make sure you have good backups |
| 22:03.47 | hesco | got that! |
| 22:03.56 | ectospasm | hesco: possible, but not easy |
| 22:04.00 | tzafrir_home | adorah, what version of Zaptel do you use? |
| 22:04.22 | adorah | <tzafrir_home>I've upgraded to the latest 1.4.9.2 |
| 22:04.30 | tzafrir_home | adorah, and anyway, that's today :-p |
| 22:04.45 | adorah | <tzafrir_home>yup:) |
| 22:05.16 | ectospasm | adorah: have you tried the svn branch of zaptel? That has a lot of fixes in it |
| 22:05.39 | tzafrir_home | ectospasm, what branch? |
| 22:05.42 | ectospasm | 1.4 branch, that is |
| 22:06.14 | tzafrir_home | I don't think there are major issues with E1/T1 cards |
| 22:06.17 | adorah | <ectospasm>No I didn't but i think I will frsh install and if it won't work that will downgrade zaptel for now |
| 22:06.32 | tzafrir_home | adorah, can you answer my questions? |
| 22:07.08 | ectospasm | adorah: the SVN branch as it stands is what will become the 1.4.10 release |
| 22:07.28 | ectospasm | so I'd go with that, it has fixes for the analog and digital cards |
| 22:08.00 | adorah | <tzafrir_home>I did ..but anyway I'm no near the machine nor have a remote access..in 12 hours will try again and if it won't work I'll report.. |
| 22:10.20 | *** join/#asterisk antlers (n=antlers@domicile.riven.net) |
| 22:11.59 | *** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 22:14.30 | antlers | forgive the question, as I'm a telephony neophyte... when someone calls into an Asterisk system, is it possible to modify the ring they receive on the remote end? |
| 22:14.35 | antlers | curious if this is even a valid concept |
| 22:18.06 | tzafrir_home | antlers, normally, no |
| 22:18.25 | tzafrir_home | not before they establish a call to Asterisk |
| 22:18.56 | *** join/#asterisk javiolo_ (n=usr@unaffiliated/javiolo) |
| 22:19.05 | tzafrir_home | It is possible to answer their call and emulate them a ring tone. But this will already be considered an established call |
| 22:19.13 | tzafrir_home | (e.g: they will be paying for it) |
| 22:19.41 | tzafrir_home | Thus the telco may actually like that you do that :-) |
| 22:20.35 | antlers | sorry, I should have been a little more clear |
| 22:20.53 | antlers | after Answer() and whilst the call is ringing to sets behind Asterisk, e.g. local to the site |
| 22:21.29 | ManxPower | antlers: show application ringing |
| 22:21.40 | ManxPower | show application playtones |
| 22:21.47 | antlers | ManxPower, I'll take that tip and go learn to fish, thanks! |
| 22:22.13 | tzafrir_home | senses one more happy telco |
| 22:22.47 | antlers | tell you this much, this time last year after moving into the new house I installed a system based on Asterisk now, I think it was |
| 22:23.04 | antlers | finally decided to rebuild my home wiring closet, new asterisk box as a virtual, etc. etc. |
| 22:23.08 | antlers | 1.6 was a kick in the face |
| 22:23.18 | antlers | going from 1.something_not_even_close_to_six |
| 22:25.14 | antlers | now to just get away from Nortel i2004 and i2007 phones and, cringe, UNISTIM |
| 22:29.44 | rickross | anyone know how to build asterisk-addons properly for valgrind? The options in "make menuselect" for asterisk are not available for asterisk-addons |
| 22:30.19 | rickross | I have NO IDEA how to set the compiler options for asterisk-addons so they are compatible with what I set in menuselect for the main asterisk |
| 22:30.50 | rickross | and I can't load asterisk under valgrind until I can get both asterisk AND asterisk-addons built with compatible compiler options |
| 22:33.07 | hesco | ~pb |
| 22:33.07 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:37.34 | hesco | OK, this is my first ever experiement with bridging a conference call with MeetMe: http://pastebin.com/d36035f75 |
| 22:38.04 | hesco | Can anyone here please advise me what's going on with this and where I might be going wrong? |
| 22:38.44 | hesco | Included is my call file, my extension.conf bridge definition and the console output. |
| 22:39.50 | tzafrir_home | hesco, what exactly do you try to bridge? |
| 22:40.02 | *** join/#asterisk BBHoss_Laptop (n=hoss@71-14-23-142.static.dctr.al.charter.com) |
| 22:40.15 | hesco | One correction, try this instead: http://pastebin.com/d7a639e96 |
| 22:40.24 | tzafrir_home | Both legs of the call end up with IAX2/diamondcard/17707551543 |
| 22:40.45 | tzafrir_home | that is: if AX2/diamondcard/1770-755-1543 is equal to that |
| 22:40.47 | hesco | tzafrir_home: I'm trying to bridge a call between my home number and my cell number. |
| 22:40.52 | BBHoss_Laptop | hey, anybody know how i can see what my externip is set to with the cli? |
| 22:41.22 | hesco | BBHoss_Laptop: as root use ifconfig |
| 22:41.40 | BBHoss_Laptop | i'm using bandwidth.com and i'm not receiving any DTMF from them. I'm pretty sure that its their problem but i want to make sure |
| 22:41.44 | tzafrir_home | actually, you don't need to be root . /sbin/ifconfig |
| 22:41.58 | BBHoss_Laptop | no i know that, i mean the externip |
| 22:42.01 | hesco | but if you have a router between you and the net, you need to run that or equivalent on your router |
| 22:42.04 | BBHoss_Laptop | thats set in sip.conf |
| 22:42.11 | BBHoss_Laptop | i want to make sure asterisk is reading it |
| 22:42.22 | antlers | when someone calls in, and the following triggers "exten => s,5,Dial(USTM/103@d_phone&USTM/102@bedroom_phone,20,r)" (e.g. ringing internal sets), where does the ring the caller hears originate from? |
| 22:44.01 | tzafrir_home | BBHoss, http://rapid.tzafrir.org.il/~tzafrir/sip_net_settings |
| 22:45.57 | hesco | tzafrir_home: ok, I changed the extension so it no longer calls the home line which is called by the call file. Now what I see in the console is at: http://pastebin.com/d326d87ef |
| 22:46.37 | hesco | Any clues on what I should try next? |
| 22:48.14 | *** join/#asterisk jmhunter (n=jacob@72.14.86.68) |
| 22:48.14 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 22:48.22 | tzafrir_home | hesco, what you do does not make sense . You first send the call into MeetMe, and then Dial elsewhere? |
| 22:49.04 | tzafrir_home | Does it actually get to anywhere on the remote side? |
| 22:49.06 | hesco | yes, I want it to dial each number in the conference call, w/o folks having to dial in. |
| 22:49.20 | hesco | It does dial my home number. |
| 22:49.24 | hesco | the 770 number |
| 22:49.57 | hesco | but I don't hear anything when I answer |
| 22:50.10 | hesco | and it does not dial my cell afterwards. |
| 22:50.30 | hesco | How do I 'send the call into MeetMe' ??? |
| 22:55.08 | hesco | . |
| 22:55.24 | hesco | Well, that is some progress, I guess. http://pastebin.com/d926460c |
| 22:56.01 | hesco | This time it prompted me to say my name then told me I was the only member of the conference, then is went quiet, until I hung up. |
| 22:56.53 | BBHoss_Laptop | tzafrir_home, still no luck on the DTMF i've done a packet capture and i can confirm there are no rfc2833 packets being sent |
| 22:58.28 | joe | what's the key combo to reboot at polycom 301? |
| 22:59.02 | bluregard | joe, for what it's worth you can reboot polycoms from the CLI |
| 23:00.21 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:01.04 | joe | bluregard: yeah, I know I just have one that isn't registered and wanted to reboot it manually |
| 23:01.30 | joe | but I can't remember the silly key combo |
| 23:01.50 | joe | nor find it on google atm |
| 23:01.55 | bluregard | joe, wish I could help you but all I have is a 501 |
| 23:02.20 | joe | bluregard: yeah msg hold + - is the combo |
| 23:02.52 | bluregard | joe, how do you like the 301 |
| 23:03.28 | joe | bluregard: they work fine just kinda limited 501s are much nicer as are the 430s |
| 23:04.21 | joe | bluregard: that combo is the 501 key combo btw |
| 23:05.51 | bluregard | joe, how does that work, hold is a softkey |
| 23:08.43 | bluregard | joe, 301 is the DND, hold and + - |
| 23:08.46 | joe | for the 301 is't volume - + hold and dnd |
| 23:09.24 | joe | bluregard: hehe, yeah just found it in the admin guide :) |
| 23:09.24 | hesco | ok, I'm heading out for food. I'll be back to learn more about MeetMe, later. Thanks folks for your help. |
| 23:09.58 | bluregard | joe, yeah, I was totally ignoring the BIG BLUE hold key when I said hold is a softkey |
| 23:11.02 | joe | bluregard: hehe :) just be careful w/ 468* it prompt for password the resets to default (sometimes) |
| 23:12.25 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
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| 23:58.10 | siya | how do I debug why a provider sends me a call on the wrong registery? |
| 23:58.45 | siya | and for some odd reason my workaround doesn't work anymore :( |
| 23:59.27 | siya | I matched the dialled number but now it just matches s in my dialplan rather than _123 (the dialled number) |