| 00:07.38 | newtonr | bunglex, what is the full dial string you set ? |
| 00:09.05 | newtonr | bunglex, just copy paste the blindxfer line from your features.conf |
| 00:10.37 | bunglex | blindxfer => ## |
| 00:11.07 | bunglex | i dont think any keypresses are being recognised |
| 00:12.09 | newtonr | if the channels are being created with Dial, are you calling it with the 't' or 'T' options to allow feature code transfers? |
| 00:12.45 | bunglex | ive tried both t and T |
| 00:13.22 | newtonr | and codes other than ## don't work either? |
| 00:13.44 | bunglex | no - and nothing shows up on the cli |
| 00:13.46 | newtonr | is this on a SIP channel? |
| 00:14.01 | newtonr | what do you mean exactly by "nothing shows up on the cli" ? |
| 00:14.43 | bunglex | i was expecting something to show up to acknowledge the feature keys had been pressed |
| 00:14.45 | danfromuk | bunglex: pastebin your logger.conf, features.conf and your extensions.conf |
| 00:15.23 | danfromuk | just the section of extensions.conf that the call goes through |
| 00:16.01 | newtonr | bunglex, it depends on what levels of verbose, debug, and dtmf messages you have turned up, also the channel technology |
| 00:16.07 | *** join/#asterisk andrewyager (~andrewyag@1.148.208.92) |
| 00:16.24 | bunglex | i see |
| 00:16.46 | danfromuk | bunglex: add dtmf to console=> in logger.conf |
| 00:16.50 | danfromuk | then reload the logger |
| 00:16.54 | newtonr | bunglex, you want verbose and debug messages going to the console (see logger.conf) and you want them turned up to level 5 (see asterisk.conf) |
| 00:17.46 | newtonr | yeah and the dtmf type that danfromuk mentioned there |
| 00:17.58 | newtonr | which doesn't have a verbosity level, its just on or off |
| 00:19.31 | newtonr | "logger show channels" on the CLI after you do what danfromuk said will help you confirm what messages are going where |
| 00:20.08 | bunglex | ok...will try now |
| 00:24.17 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.58) |
| 00:27.29 | bunglex | great - ok now i see the DTMF in the cli |
| 00:29.07 | bunglex | begin '#' recieved on sip and then dtmf begin passthrough '#' on sip.. end accepted etc. |
| 00:30.50 | bunglex | i had put include => parkedcalls in sip.conf - is that correct? i cant see any other reference to parkedcalls in sip.conf |
| 00:32.26 | bunglex | http://pastebin.com/NfC5LJ8W features.conf - which seems as most of it is commented out |
| 00:32.26 | danfromuk | put that in extensions.conf |
| 00:32.33 | danfromuk | not sip.conf |
| 00:33.02 | bunglex | sorry i meant extensions.conf |
| 00:33.16 | danfromuk | It may not be being applied to that sip peer's dial plan |
| 00:33.23 | *** join/#asterisk gartral (~gartral@unaffiliated/gartral) |
| 00:33.41 | danfromuk | can you pb your entire extensions.conf? |
| 00:34.33 | Penguin | I give my phones a context of "phones" and the phones context includes parkedcalls. |
| 00:37.06 | bunglex | ive included parkedcalls - but cant see where the parked calls context is defined in my extensions.conf.......http://pastebin.com/z7KqRSCp |
| 00:37.27 | Penguin | Did you run "dialplan reload" after you changed and saved extensions.conf? |
| 00:37.45 | Penguin | You always have to run dialplan reload when changing extensions.conf. |
| 00:37.50 | bunglex | yes, welll core stop now and reloaded |
| 00:38.12 | Penguin | That's a waste. dialplan reload is the appropriate thing to reload extensions.conf. |
| 00:39.02 | bunglex | ahh ok - thanks... its a steep curve for me |
| 00:39.27 | Penguin | At some point, you'll come over the top. |
| 00:39.31 | *** join/#asterisk suneye (~atcmmi@116.25.196.132) |
| 00:39.44 | ChannelZ-Wk | and then we'll all crash and die at the bottom |
| 00:41.27 | bunglex | god i hope so - but jesus it seems like a massive massive mountain.....head wrecker! im using 3cx softphone - when i open another line i hear the MOH - but i just cant figure out hot to put on hold with keypresses (that i need for handsets) |
| 00:42.08 | Penguin | The phones don't have hold buttons? |
| 00:42.23 | m0sphere | i'm using callcentric DID for inbound calls. i then have DID forwarding to an extension, which is set up and registered to my freepbx server. I'm calling my DID from cellphone1 and reaching cellphone2, then with cellphone3 i call the same DID and it just rings. I assume this is because my DID is forwarding to an extension that is already on a call. How do I set this up to accept more calls? |
| 00:42.52 | bunglex | nope - no hold button |
| 00:44.08 | runfromnowhere | Hmm, has anyone ever seen Asterisk 11 make a database entry for a call without mentioning it at all in the verbose log? I'm searching for log data on this call via UniqueID but there's no mention....even though it's been assigned a "recordingfile" filename |
| 00:44.33 | ChannelZ-Wk | m0sphere: There's like 5 possible reasons, freepbx could be doing something (or nothing as it were), callcentric might be doing something... |
| 00:45.05 | danfromuk | bunglex: what context do the phones have? |
| 00:47.11 | m0sphere | i don't believe it's freepbx because the cli doesn't show cellphone3 ever reaching it |
| 00:47.25 | ChannelZ-Wk | then it'd be callcentric |
| 00:47.26 | danfromuk | bunglex: to park a call, you need to press the transfer button, and transfer the call to the call parking extension |
| 00:49.32 | bunglex | context 'main' - one of the phones does not have a transfer button |
| 00:50.19 | danfromuk | features.conf allows you to add digits for transfer |
| 00:50.25 | ChannelZ-Wk | features.conf then :/ |
| 00:50.26 | ChannelZ-Wk | dtmf transfer |
| 00:50.34 | Penguin | If you need to do dtmf transfers, specify the t dial option on the phone's extension. |
| 00:50.54 | danfromuk | atxfer => *2 |
| 00:50.58 | danfromuk | What does *2 do? |
| 00:51.30 | Penguin | It's usually just the # to trigger it, I believe. But you have to have the t dial option on your extension when calls go to you, and you have to use the T dial option on extensions when you call out from your phone. |
| 00:51.54 | danfromuk | Penguin: he pastebin'd his features.conf and hes got *2 for attended transfer |
| 00:52.01 | Penguin | I didn't see it. |
| 00:52.08 | Penguin | I just look in features.conf sample. |
| 00:52.20 | Penguin | blind transfer is # by default. |
| 00:52.38 | ChannelZ-Wk | #1 |
| 00:52.50 | danfromuk | Yes, but to park a call and hear the parked extension number, would he not have to do an attended transfer? |
| 00:52.56 | Penguin | You can use #1 if you want, but it is just # by default. |
| 00:52.57 | ChannelZ-Wk | err # if not set, samples uses #1 |
| 00:53.00 | runfromnowhere | This almost looks like a bug... |
| 00:53.22 | bunglex | the extension is exten => 6002,1,Dial(Sip/BoxxPC,15,T) - |
| 00:53.42 | danfromuk | try tT for testing and see what happens |
| 00:53.58 | Penguin | If you call extension 6002, when BoxxPC answers, BoxxPC should be able to hit the feature code on the keypad and it will work. |
| 00:54.01 | danfromuk | Dont forget to remove the one you dont need |
| 00:54.18 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 00:57.15 | bunglex | wow the tT worked!!!! |
| 00:57.36 | *** join/#asterisk felipealmeida (~user@177.41.16.10) |
| 00:57.40 | ChannelZ-Wk | You probably only want t |
| 00:57.48 | danfromuk | one allows the caller to transfer, the other allows the callee to transfer. |
| 00:57.52 | Penguin | Depends on which way the call is going. |
| 00:58.03 | ChannelZ-Wk | yes |
| 00:58.07 | Penguin | If someone called him at 6002, and he wants to transfer, he'd want T. |
| 00:58.28 | Penguin | No, I think I said that backward. |
| 00:58.29 | ChannelZ-Wk | mm, no that'd be t. |
| 00:58.34 | *** join/#asterisk Smirker (~x@CPE-144-137-156-146.lnse3.cha.bigpond.net.au) |
| 00:58.39 | danfromuk | Penguin: yes, but which one is he trying to transfer from. Which end of the call is trying the transfer |
| 00:58.49 | Penguin | Yeah, I can't erase after I press Enter. |
| 00:59.01 | ChannelZ-Wk | Chances are you want t on your "incoming" extensions and T on your "outgoing" |
| 00:59.09 | bunglex | i am trying to transfer an incoming call |
| 00:59.20 | danfromuk | I use t |
| 00:59.30 | Penguin | For calls coming to you, the dial that rings your phone needs the t so you can transfer. |
| 00:59.40 | Smirker | Hi guys. Unfortunately I'm working on an old production server running Asterisk 1.6, and I need to access something similar to MASTER_CHANNEL(), to update MASTER_CHANNEL(CHANNEL(musicclass)). Is there any way to set the musicclass on the master channel in Asterisk 1.6? |
| 00:59.54 | danfromuk | Right. I'm off for the night. Glad I could help for a change! |
| 01:00.07 | bunglex | thank you thank you thank you :-) |
| 01:00.08 | ChannelZ-Wk | And then get a phone with a transfer button :) |
| 01:00.09 | Smirker | My aim is to implement one MOH for pre-answer, and then one MOH for post-answer and they put a customer on hold. |
| 01:00.52 | Penguin | I don't have any idea what a master channel is. |
| 01:00.59 | danfromuk | bunglex: good luck. A little tip Ive discovered.... include a pastebin of your code in your question. More chance of someone answering. |
| 01:01.24 | bunglex | for gods sake next problem is that the DTMF tone works on softphones - but for some reason it doesnt on the SPA :-) thats for tomorrow. |
| 01:01.41 | bunglex | will do...thanks danfromuk |
| 01:01.46 | Penguin | SPA what? |
| 01:01.49 | ChannelZ-Wk | Old way is SetMusicOnHold(classname) |
| 01:01.55 | bunglex | SPA3102 |
| 01:02.01 | ChannelZ-Wk | But I'm not sure when old became old |
| 01:02.08 | Penguin | It can do transfers natively. |
| 01:02.11 | Penguin | No DTMF required. |
| 01:02.43 | danfromuk | Look at the dtmfmode for that peer. The SPA3102 might not be sending what asterisk is expecting. |
| 01:02.57 | bunglex | ahh good point.... actually it must be the handset then |
| 01:02.59 | Smirker | ChannelZ-Wk, I need to change the MOH after the Queue is answered (via a gosub or macro). Problem is those calls execute on the called channel, and if I change the MOH it changes it on the wrong channel. Hence I need to use MASTER_CHANNEL, but it's not available on 1.6. :'( |
| 01:03.15 | Penguin | Set(CHANNEL(musicclass)=native) |
| 01:03.37 | danfromuk | There are different ways that dtmf is transmitting. It can be within the sip data, or within the audio itself. I use rfc2833 and not had a problem. |
| 01:04.18 | Penguin | I always use rfc2833 as well. |
| 01:04.56 | Penguin | But to transfer using an SPA-3102 with a phone hooked up to it, you don't need to do DTMF transfers because the ATA can transfer. |
| 01:05.32 | ChannelZ-Wk | By thought power? |
| 01:05.43 | Penguin | hook flash |
| 01:05.59 | bunglex | no sorry im getting confused - the phone that ## isnt working on is an old analogue one, but its actually connected to the router as a sip client thats then connected to asterisk. |
| 01:06.15 | bunglex | the spa is just forwarding all incoming calls - so it cant be that |
| 01:06.45 | Penguin | Forwarding? There's a service code for that. |
| 01:07.07 | ChannelZ-Wk | Now I'm confused, you're using the analog line portion or the analog phone portion? (or perhaps both) |
| 01:08.26 | bunglex | i have an old analogue phone connected to a router that supports sip and that port registers to asterisk. |
| 01:08.42 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 01:09.33 | ChannelZ-Wk | The router being the 3102 or is there now some other device involved? |
| 01:09.55 | bunglex | no its a vigor 2850 |
| 01:12.07 | bunglex | and i see it has DTMF relay |
| 01:15.10 | bunglex | i changed it to outbound and now get RTP rear too short in asterisk cli |
| 01:15.52 | bunglex | outbound has rfc2833 next to it :) |
| 01:16.01 | citywok | how goes it? |
| 01:16.11 | ChannelZ-Wk | goes home |
| 01:16.13 | Penguin | Make sure the sip entry for that device is set to rfc2833. |
| 01:16.20 | citywok | yea... just about that time for me :p |
| 01:19.04 | bunglex | thank you for everyones help - appreciated |
| 01:23.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.94) |
| 01:23.59 | *** join/#asterisk mitchrodrigues (~mitchrodr@184.250.224.209) |
| 01:32.36 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 01:35.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:35.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:36.02 | *** join/#asterisk mitchrodrigues (~mitchrodr@184.250.224.209) |
| 01:41.35 | *** join/#asterisk suneye (~atcmmi@116.25.196.132) |
| 01:58.07 | *** join/#asterisk CrashHD (~na@204-195-127-247.wavecable.com) |
| 02:10.40 | *** join/#asterisk SushiB (~Thunderbi@187.184.23.221.cable.dyn.cableonline.com.mx) |
| 02:22.57 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 02:24.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.67) |
| 02:24.34 | *** join/#asterisk andrewyager (~andrewyag@1.148.236.240) |
| 02:28.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:43.26 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 02:45.23 | *** join/#asterisk andrewyager (~andrewyag@11-104-141-114.static-dsl.realworld.net.au) |
| 02:46.39 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 02:48.04 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 02:54.35 | *** join/#asterisk andrewyager (~andrewyag@11-104-141-114.static-dsl.realworld.net.au) |
| 03:24.23 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.55) |
| 03:42.46 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 03:45.36 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 03:47.53 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 03:48.31 | *** join/#asterisk endemic (~endemic@lynx.onvox.net) |
| 03:48.37 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 03:51.42 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 03:59.24 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 04:02.03 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 04:02.11 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:06.03 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 04:07.13 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 04:07.51 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 04:08.08 | *** join/#asterisk jblack (~jblack@70-91-203-214-BusName-Washington.hfc.comcastbusiness.net) |
| 04:08.36 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 04:10.11 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 04:11.54 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 04:12.59 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 04:13.06 | saliak | I'm trying to get the BLF on a GXP2124 to work. I've configured it per the GXP2000 instructions provided by grand stream (setting up a context with hint for the extensions, and adding them to my sip config as subscribecontext with call-limit and limitonpeers). the lamps are green by default, but never change state (even if the other extensions are on the phone). that isn't the correct behavior, eh? can anyone tell me what it's supposed t |
| 04:13.09 | saliak | (I'd assume be red to indicate that the extension is in use, or maybe flash if it's ringing?), and what might be going on? |
| 04:15.14 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 04:16.55 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 04:22.37 | *** join/#asterisk jblack (~jblack@m9a2036d0.tmodns.net) |
| 04:24.06 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.52) |
| 04:32.36 | *** join/#asterisk andrewyager (~andrewyag@11-104-141-114.static-dsl.realworld.net.au) |
| 04:35.43 | ChannelZ | I don't know anything about your phone, but one place to start is are you even seeing the subscription from the phone in asterisk? (core show hints, sip show subscriptions) |
| 04:38.45 | saliak | yeah, i see the watcher and the active subscriptions |
| 04:40.07 | ChannelZ | And presumably with verbose you see it changing the hint states and notifying the device on the console |
| 04:41.35 | *** join/#asterisk atcmmi (~atcmmi@116.25.197.246) |
| 04:42.05 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 04:43.12 | *** join/#asterisk jploh (~textual@49.144.120.20) |
| 04:43.57 | saliak | ChannelZ: ah, ok. yeah, so status never changes from Idle when i do core show hints, even when i make a call |
| 04:46.09 | ChannelZ | Hmmm |
| 04:48.26 | ChannelZ | So I guess next is what do your hints look like? |
| 04:49.21 | saliak | ChannelZ: in the dial plan? or when i core show hints? |
| 04:49.54 | ChannelZ | either is fine I suppose |
| 04:50.51 | ChannelZ | pb your core show hints and sip show subscriptions |
| 04:51.42 | saliak | http://pastebin.com/mhc4GjFx |
| 04:51.50 | saliak | oops, sorry, need to add some more to it |
| 04:53.19 | saliak | ok, http://pastebin.com/ACAfS5Td |
| 04:55.38 | saliak | When it says the extension it's watching is 8000@blf_kiinde, should that actually be 800@voipms-outbound (the default context for 8000)? |
| 04:56.20 | ChannelZ | no it's the hint and hint context |
| 04:56.59 | saliak | OK. strange that the state never changes. It does have the "unavailable" ones right, though |
| 04:57.01 | ChannelZ | So that all looks right near as I can tell. But you say when you make a call on any of those phones (SIP/8000, 8001, etc) none of their states change in 'sip show hints'? |
| 04:57.10 | Penguin | It means hint context of blf_kiinde, hint extension of 8000, and you are watching the device SIP/8000. |
| 04:57.51 | saliak | ChannelZ: right. it says idle even if 8000 is on a call |
| 04:58.12 | Penguin | Did you enable call counters? |
| 04:58.30 | ChannelZ | yeah was just going to ask that |
| 04:58.41 | saliak | Penguin: probably not |
| 04:58.47 | Penguin | callcounter=yes <--- |
| 04:58.54 | ChannelZ | callcounter=yes in [general] |
| 04:59.18 | saliak | i have limitonpeers=yes |
| 04:59.22 | saliak | but no callcounter |
| 04:59.24 | saliak | let me try that |
| 04:59.49 | ChannelZ | PREPARE FOR BLINKYLIGHTS! |
| 05:00.14 | saliak | nailed it |
| 05:00.30 | saliak | hat tip to you both, as always |
| 05:01.18 | ChannelZ | Da! Das blinkenlight |
| 05:01.37 | saliak | just red so far. testing blinking |
| 05:02.35 | saliak | and it blinks! |
| 05:04.22 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:07.50 | ChannelZ | has a seizure |
| 05:08.00 | saliak | :) |
| 05:24.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.52) |
| 05:36.19 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 05:41.39 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 05:44.17 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 06:10.54 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 06:15.55 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:21.52 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 06:24.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.72) |
| 06:34.48 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 06:51.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.72) |
| 06:58.56 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 07:02.53 | *** join/#asterisk andrewyager (~andrewyag@11-104-141-114.static-dsl.realworld.net.au) |
| 07:06.57 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 07:15.23 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.72) |
| 07:19.59 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:30.41 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 07:48.30 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:52.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.72) |
| 08:02.07 | *** join/#asterisk jblack (~jblack@70-91-203-214-BusName-Washington.hfc.comcastbusiness.net) |
| 08:08.21 | *** join/#asterisk D30 (~deo@58.71.19.178) |
| 08:08.31 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:17.32 | D30 | hi all, have you experience some echo when you are trying to call ? |
| 08:18.14 | D30 | im using tdm800p card from digium |
| 08:18.33 | D30 | i can see the echo cancellation is loaded but theres still echo when calling :( |
| 08:19.35 | *** join/#asterisk andrewyager (~andrewyag@11-104-141-114.static-dsl.realworld.net.au) |
| 08:19.39 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 08:31.46 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 08:32.55 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 08:32.58 | wasanzy | hello |
| 08:33.35 | wasanzy | I hv tried looking for how to install codec_slin and codec_slin44 but couldn't find a way out |
| 08:33.43 | wasanzy | any help? |
| 08:37.30 | *** join/#asterisk esaym153 (~esaym153@216-45-91-132.gvec.net) |
| 08:45.03 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:13.20 | *** join/#asterisk martinfletcher_ (~martinfle@87.237.70.109) |
| 09:14.27 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:19.49 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:21.10 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 09:24.22 | *** join/#asterisk esaym153 (~esaym153@216-45-91-132.gvec.net) |
| 09:28.18 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:c03c:3a7e:3a8e:398e) |
| 09:31.18 | *** join/#asterisk andrewyager (~andrewyag@11-104-141-114.static-dsl.realworld.net.au) |
| 09:39.44 | *** join/#asterisk magespawn (~magespawn@105-237-58-254.access.mtnbusiness.co.za) |
| 09:49.05 | *** join/#asterisk esaym153 (~esaym153@216-45-91-132.gvec.net) |
| 09:56.01 | *** join/#asterisk suneye (~atcmmi@121.34.41.65) |
| 10:00.02 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 10:00.43 | wasanzy | any asistant? |
| 10:10.31 | kaldemar | ~ask |
| 10:10.31 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:18.18 | *** join/#asterisk smirker (~x@CPE-144-137-156-146.lnse3.cha.bigpond.net.au) |
| 10:28.36 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 10:32.00 | MaliutaLap | assistants? in here? You really think we make _that_ much money that we have assistants to answer questions for us? ;) |
| 10:32.28 | MaliutaLap | but then again, there is always infobot |
| 11:19.55 | *** join/#asterisk andrewyager (~andrewyag@11-104-141-114.static-dsl.realworld.net.au) |
| 11:26.19 | *** join/#asterisk eirirs_ (~eirirs@snowyinn.nsa.org) |
| 11:33.37 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
| 11:49.51 | rox | If I have a number of SIP friends configured into a callgroup/pickupgroup, how can I configure the office phone LED to light up, when an interceptable call comes? |
| 11:50.53 | rox | For example, I have office phones A, B and C configured into a callgroup/pickupgroup, when a call comes to phone A, phone A is ringing, but how do i get phones B and C to flash their LEDs? |
| 11:51.10 | rox | do I need hint for that? |
| 11:58.03 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
| 12:02.19 | mirela666 | rox: yes a hint can do that |
| 12:03.38 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 12:03.41 | wasanzy | hello |
| 12:03.50 | wasanzy | any help please? |
| 12:05.53 | Chainsaw | I can't help you, because you haven't asked any questions. |
| 12:13.29 | wasanzy | Chainsaw: I asked a questions long time but let me just ask again |
| 12:13.59 | wasanzy | I hv tried looking for how to install codec_slin and codec_slin44 but couldn't find a way out |
| 12:17.30 | rox | mirela666: thank you very much, I think I have solved my problem :-) |
| 12:21.45 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 12:27.27 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 12:28.32 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 12:30.19 | kaldemar | wasanzy: as a better question. where did you come up with those codecs? |
| 12:30.35 | kaldemar | s/as/ask/ |
| 12:31.07 | kaldemar | hmpf. |
| 12:44.36 | wasanzy | kaldemar: I was ask to install it by my boss |
| 12:46.03 | mirela666 | wasanzy: core show codecs, would 100% show you that you have slin |
| 12:46.14 | mirela666 | <PROTECTED> |
| 12:47.24 | wasanzy | how do I check if they are working or not? |
| 12:48.18 | mirela666 | play any .wav file and in cli you will see something like "Playing xxx.slin on channel xxx" |
| 12:48.48 | mirela666 | * converts wav to slin before playback |
| 12:48.50 | mirela666 | I think |
| 12:48.53 | wasanzy | ok |
| 12:49.27 | mirela666 | but better wait for someone more experianced and with better answer |
| 13:10.29 | wasanzy | the codecs are not in the module directory, so where can I download it? |
| 13:15.33 | file | there are no codec modules for signed linear |
| 13:16.16 | wasanzy | file: so I can't compile it? |
| 13:16.31 | file | you can't compile codec modules that don't exist |
| 13:17.07 | wasanzy | does it mean there is no codec signed linear? |
| 13:17.21 | file | it means there is no codec module for it |
| 13:17.28 | file | but internally other codecs transcode to and from signed linear |
| 13:18.26 | wasanzy | am a bit confuse, if I want to use codec_slin, I can't use it? |
| 13:18.44 | file | that module doesn't exist... so what are you trying to do? |
| 13:19.35 | wasanzy | when I did core codecs show, I see codec_slin and codec_slin44 listed so what does it mean? |
| 13:20.13 | file | that are codecs that Asterisk knows about |
| 13:20.17 | file | not modules that are loaded |
| 13:20.36 | file | internally Asterisk knows about signed linear, but it does not have a codec_slin module because it would do nothing |
| 13:21.41 | wasanzy | ok so if say I have an audio that needs signed linear codec, what happens? |
| 13:21.56 | file | define "needs signed linear codec" |
| 13:22.12 | file | other modules will transcode to and from signed linear, or signed linear can be provided as-is from elsewhere |
| 13:22.52 | wasanzy | good, so it means I can actually get the codec from somewhere else? |
| 13:23.02 | file | get what codec? |
| 13:23.10 | kaldemar | wasanzy: do you know what a codec is? |
| 13:23.43 | file | goes to catch a bus |
| 13:24.20 | wasanzy | kaldemar: yes I know what a codec is |
| 13:24.46 | wasanzy | am just wondering why my boss asked me to install it if it is not something installable |
| 13:27.41 | Chainsaw | wasanzy: It was a trick question, clearly. |
| 13:27.51 | Chainsaw | wasanzy: You may have failed that test :( |
| 13:28.36 | wasanzy | interesting |
| 13:29.13 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 13:29.23 | MaliutaLap | wasanzy: did he send you out for left handed screwdrivers and plaid paint too? ;) |
| 13:30.24 | wasanzy | MaliutaLap: hheheheh no |
| 13:30.41 | MaliutaLap | wasanzy: you say that _now_ :P |
| 13:31.04 | *** join/#asterisk rolek (~rolek@d57d3382.static.ziggozakelijk.nl) |
| 13:31.08 | *** join/#asterisk Draecos (~Draecos@106-69-19-92.dyn.iinet.net.au) |
| 13:31.09 | *** join/#asterisk insha (~insha@24.31.225.204) |
| 13:31.21 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 13:31.24 | *** part/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 13:31.53 | MaliutaLap | wasanzy: haven't you learnt yet that when "the boss" asks you to "install" something, then you just put it on the project list and claim you're doing "research" to get it to work properly ... then get on with the real work |
| 13:32.01 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 13:32.02 | MaliutaLap | bosses are as bad as devs |
| 13:33.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:34.59 | *** join/#asterisk insha (~insha@24.31.225.204) |
| 13:36.29 | wasanzy | MaliutaLap: Wow am not aware of that |
| 13:37.50 | rolek | Hi everyone |
| 13:38.03 | rolek | Does anyone know if there is a dialplan variable or function that will tell me i |
| 13:38.05 | rolek | f a jitterbuffer is enabled for a specific peer? |
| 13:39.04 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
| 13:40.25 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:41.21 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 13:49.45 | file | hi. |
| 13:53.28 | Chainsaw | Hello. |
| 13:58.02 | *** join/#asterisk tty-1 (~jumperswi@ltea-047-067-204-221.pools.arcor-ip.net) |
| 13:58.05 | tty-1 | hi |
| 13:58.32 | tty-1 | I want to know if a single 'normal' SIP number could receive multiple ingoing calls if the SIP client supports this (asterisk does?)? |
| 13:58.49 | tty-1 | Something like ISDN, so that multiple callers can reach one single Asterisk SIP server. |
| 13:59.24 | Chainsaw | Yes, it can handle concurrent calls. |
| 13:59.49 | Chainsaw | If you understand ISDN already, consider SIP like the D-channel and RTP like the B-channels. |
| 14:00.12 | Chainsaw | Provided you have somewhere to route the audio streams, you can accept as many calls as you like. Until you saturate your connectivity. |
| 14:00.23 | Chainsaw | (And beyond, if you're really mad) |
| 14:00.31 | *** join/#asterisk serafie (~erin@nat/digium/x-dtzrmqjurfpraexs) |
| 14:01.33 | [TK]D-Fender | tty-1: Thre is no such thing as a "SIP number". |
| 14:02.23 | [TK]D-Fender | tty-1: A DID arrives at your telco (or ITSP in this case), and they can deliver it to you via SIP for as many incoming channels as your plan and bandwidth allows for. |
| 14:02.31 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:02.32 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:02.39 | tty-1 | so many new abbreviations :) |
| 14:03.01 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 14:03.43 | tty-1 | [TK]D-Fender: OK, so it goes over ARCOR, do they support this, too? Is it called SIP trunking or is this something different? |
| 14:03.53 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:04.02 | tty-1 | [TK]D-Fender: so for each incoming call the normal costs are billed (normally), right? |
| 14:06.49 | [TK]D-Fender | ~itsp |
| 14:06.50 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 14:07.10 | [TK]D-Fender | tty-1: Each call is just a call. IT costs whatever your plan with your provider says it does. |
| 14:07.55 | [TK]D-Fender | tty-1: Some charge per channel (or extra once you pass a certain point), som simply charge per minute for however many channels you need, etc |
| 14:08.00 | [TK]D-Fender | ~did |
| 14:08.00 | infobot | well, did is Direct Inward Dialing, or just a phone number |
| 14:09.14 | [TK]D-Fender | tty-1: ARCOR appears to be an ITSP and that means they deliver calls to you via a VoIP protocol, which from having seen them before, is SIP |
| 14:19.52 | tty-1 | [TK]D-Fender: What I want to understand is the concept of extension in Asterisk, it is not a plugin or a phone number extension but rather some kind of trunk or endpoint, right? |
| 14:20.31 | [TK]D-Fender | tty-1: Are you asking what ARCOR is in relation to Asterisk? |
| 14:21.00 | tty-1 | [TK]D-Fender: somehow, I want that asterisk server accept incoming arcor sip calls |
| 14:21.35 | [TK]D-Fender | Yes, that is normal |
| 14:21.38 | [TK]D-Fender | And certainly can |
| 14:21.53 | [TK]D-Fender | I've seen several people come in here using them |
| 14:22.30 | [TK]D-Fender | With Asterisk, it is all just SIP. |
| 14:23.00 | [TK]D-Fender | there is no real difference between a call from them ,or any other provider, or a SIP hardphone, softphone, etc |
| 14:23.12 | [TK]D-Fender | all of the processing difference is up to you and your dialplan |
| 14:31.51 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:40.45 | *** join/#asterisk dxd828 (~dxd828@109.144.246.107) |
| 14:42.33 | *** join/#asterisk felipealmeida (~user@177.41.16.10) |
| 14:52.59 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:53.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:54.14 | tty-1 | [TK]D-Fender: when there is a single number and a single endpoint (one answer box), then only one asterisk extension is needed. |
| 14:55.45 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:00.39 | *** join/#asterisk rogers (rogers@bling.bling.org) |
| 15:02.05 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 15:06.27 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:08.12 | *** join/#asterisk navaismo (~navaismo@189.241.101.4) |
| 15:15.24 | *** join/#asterisk dpeloquin (~uid13057@gateway/web/irccloud.com/x-xpduoukkqgfpdjed) |
| 15:16.01 | *** join/#asterisk SushiB (~Thunderbi@200.77.217.106) |
| 15:17.43 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 15:21.17 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 15:25.38 | *** join/#asterisk Draecos (~Draecos@106-69-19-92.dyn.iinet.net.au) |
| 15:37.24 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 15:38.12 | [TK]D-Fender | tty-1: What do you mean by "extension" and "answer box"? |
| 15:38.33 | *** join/#asterisk martinfletcher_ (~martinfle@87.237.70.109) |
| 15:39.27 | tty-1 | [TK]D-Fender: there is one single sip phone number (that abbreviation) and now asterisk should listen on incoming calls and route them to its integrated answer box. |
| 15:40.28 | [TK]D-Fender | tty-1: Asterisk needs a sip.conf entry to match them and tell them what context in the dialplan to place the call into. |
| 15:40.57 | [TK]D-Fender | tty-1: That call will come in targeting a # for which you need an extension to match, and then what you do is up to you. |
| 15:41.20 | [TK]D-Fender | tty-1: Vocemail() is it's own dialplan application and needs you to have configured a box in it, etc |
| 15:41.25 | Penguin | [TK]D-Fender: its |
| 15:42.14 | [TK]D-Fender | Was not was :) |
| 15:42.15 | navaismo | Penguin, its != it's ? it's = it is? |
| 15:42.27 | [TK]D-Fender | vanCorrect |
| 15:42.39 | tty-1 | [TK]D-Fender: sip.conf [general] register and [arcor] auth is there |
| 15:42.57 | [TK]D-Fender | I type fast and think little these days so I end up writing it wrong regardless of knowing the rule perfectly :) |
| 15:43.06 | [TK]D-Fender | Chalk it up to lazy & sloppy |
| 15:43.19 | [TK]D-Fender | tty-1: extensions.conf = all call processing. |
| 15:43.33 | tty-1 | [TK]D-Fender: [incoming] |
| 15:43.34 | tty-1 | exten => 01234567890,1,Dial(SIP/30) |
| 15:43.36 | tty-1 | exten => 01234567890,2,VoiceMail(2000,u) |
| 15:43.45 | tty-1 | (I removed the phone numbers ;) ) |
| 15:43.47 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
| 15:43.54 | [TK]D-Fender | tty-1: PASTEBIN... don't flood in-channel |
| 15:43.56 | [TK]D-Fender | ~pb |
| 15:43.56 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:43.58 | tty-1 | sorry |
| 15:44.14 | tty-1 | but there is so few traffic, would three lines be so bad? |
| 15:44.25 | [TK]D-Fender | tty-1: Keep in mind |
| 15:44.56 | [TK]D-Fender | tty-1: anyway, looks like that will dial 1 device, and then fall to voicemail if the device rejects it or is unable to be called |
| 15:45.16 | [TK]D-Fender | tty-1: Assumiing that that exten matches what they are looking for |
| 15:48.03 | tty-1 | [TK]D-Fender: so I don't need the dial entry if I only want the answer box |
| 15:48.09 | tty-1 | [TK]D-Fender: on the other hand... |
| 15:48.53 | tty-1 | [TK]D-Fender: There is SIP phone in a building and a remote Server which runs asterisk |
| 15:49.01 | tty-1 | [TK]D-Fender: one SIP phone |
| 15:50.45 | tty-1 | [TK]D-Fender: I want to test the configuration. Can I also make a test / dummy call using the CLI (shell)? |
| 15:51.05 | [TK]D-Fender | use your sip phone to test |
| 15:51.28 | [TK]D-Fender | basically... the only thing it will prove is if you set up that voicemail box... |
| 15:51.56 | [TK]D-Fender | because ti won't prove that arcor's call is arriving, getting accepted, is pointed to the right context, or is targeting the exten you defined |
| 15:52.59 | tty-1 | [TK]D-Fender: Can I also call a 'normal' phone number using the cli? |
| 15:53.12 | tty-1 | [TK]D-Fender: just to ring, no sound necessary. I want to assure that the call gets through. |
| 15:53.43 | [TK]D-Fender | tty-1: That would be using * CLI as a "phone" which means you'd need chan_alsa to be configured. |
| 15:53.52 | [TK]D-Fender | "help console dial" <- |
| 15:54.00 | [TK]D-Fender | and you'd need dialplan to dial out. |
| 15:54.04 | [TK]D-Fender | that is separate... |
| 15:55.30 | *** join/#asterisk gerritfromsa (~androirc@8ta-151-117-229.telkomadsl.co.za) |
| 15:57.05 | *** part/#asterisk rox (~rox@mx.abraxas.si) |
| 15:57.10 | tty-1 | [TK]D-Fender: also just for a simple test call? |
| 15:57.26 | tty-1 | [TK]D-Fender: so I would have to install alsa on that server, too, right? |
| 15:57.50 | [TK]D-Fender | Set up a sofphone or your SIP phone |
| 15:58.03 | [TK]D-Fender | not worth the trouble to use the console. |
| 15:58.57 | tty-1 | [TK]D-Fender: Where can I check if asterisk server is able to connect to the arcor proxy and start receiving calls? |
| 15:59.13 | [TK]D-Fender | To test receiving.. you have to just CALL. |
| 15:59.27 | [TK]D-Fender | "sip show registry" will show if you've registered to them or not |
| 15:59.41 | [TK]D-Fender | usually a good start to knowing if they even know where to send your calls |
| 16:00.14 | tty-1 | [TK]D-Fender: and how are sip 'endpoints' prioritized? There is a router far away from that server running asterisk which provides sip-analog phoning. How can I assure that the call goes to the asterisk server first, then to the router (and the phone on it)? |
| 16:01.03 | [TK]D-Fender | tty-1: I have no idea what that other box is doing |
| 16:01.35 | [TK]D-Fender | There are a number of NAT related settings in sip.conf you have to do, including forward SIP & RTP to * if your server is behind NAT |
| 16:02.47 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 16:04.16 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:09.58 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 16:10.24 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-tetgsswfitrgmvqe) |
| 16:10.28 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-qzynxdnpoagfqirm) |
| 16:10.29 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:10.30 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 16:11.27 | *** join/#asterisk SushiB1 (~Thunderbi@201.170.28.99.dsl.dyn.telnor.net) |
| 16:18.03 | tty-1 | [TK]D-Fender: ok, I added the config, started asterisk and opened the cli |
| 16:18.18 | tty-1 | [TK]D-Fender: where can I get the error messages if asterisk fails to connect to arcor? |
| 16:21.49 | [TK]D-Fender | Do you see it saying "registered"? |
| 16:22.07 | [TK]D-Fender | Did you configure all of the NAT settings for [general] as specified in the sample config? |
| 16:25.28 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 16:35.01 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 16:35.17 | tty-1 | [TK]D-Fender: the server is a Vserver without any NAT infront |
| 16:36.14 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:37.29 | *** join/#asterisk SushiB (~Thunderbi@200.77.217.106) |
| 16:38.18 | [TK]D-Fender | tty-1: So * has a PUBLIC IP on it? |
| 16:39.30 | tty-1 | [TK]D-Fender: there is no forwarding at all in asterisk.conf |
| 16:41.01 | [TK]D-Fender | asterisk.conf has nothing to do with "forwarding" |
| 16:41.06 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 16:41.12 | [TK]D-Fender | I just asked you what kind of IP your server had on it |
| 16:41.39 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 16:41.49 | tty-1 | [TK]D-Fender: yes, it is a public ip |
| 16:42.27 | [TK]D-Fender | ok, so what is this other router you're talking about? |
| 16:42.49 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:44.01 | tty-1 | [TK]D-Fender: the router is somehwere else and provides phoning over SIP to an analog phone |
| 16:44.34 | tty-1 | [TK]D-Fender: the point is: Who gets the incoming call first / at all? The router or the vserver (with asterisk)? |
| 16:44.36 | *** join/#asterisk emper0r (~tony@galileo.red.sld.cu) |
| 16:44.40 | emper0r | hi... |
| 16:44.57 | [TK]D-Fender | tty-1: Why would this other router get the call? |
| 16:45.04 | emper0r | i'm cont getting trouble connecting with trunk to skype.. |
| 16:45.15 | emper0r | somebody have some trunk with skype working.. ? |
| 16:45.19 | tty-1 | [TK]D-Fender: because currently all incoming calls are going to the router and from it to the analog phone |
| 16:45.23 | [TK]D-Fender | tty-1: You are providing no details about how it is involved in the routing path, what service it is configured for use witgh, etc |
| 16:45.30 | tty-1 | [TK]D-Fender: hm, right |
| 16:45.46 | emper0r | i need check some network testing to confirm or have more details to fix my asterisk |
| 16:46.51 | *** join/#asterisk CeBe (~CeBe@port-92-206-212-107.dynamic.qsc.de) |
| 16:47.08 | *** join/#asterisk outtolunc (~me@c-67-170-214-55.hsd1.ca.comcast.net) |
| 16:52.41 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:01.59 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 17:07.45 | cusco | hey, if telco asks if pri card is as TE or NT |
| 17:07.49 | cusco | how do I know? |
| 17:07.57 | cusco | I'm guessting it is as a slave to their timing source |
| 17:08.05 | cusco | but not sure if that is what theymean |
| 17:09.20 | Penguin | navaismo: Yes, that's right. it's = it is. "its" is the possessive pronoun indicating that something belongs to it, like its own dialplan. |
| 17:12.31 | outtolunc | cusco: timing is usually a config setting on the digium/sangoma cards.. usually you let the carrier provide the timing on PRI (T1 w/isdn) circuits so if you were going to provide the timing from your end (you would act as the NT) otherwise you would accept the timing (acting as TE) |
| 17:14.32 | navaismo | thx Penguin |
| 17:17.26 | *** join/#asterisk lvlinux (~n1gg@adsl-98-86-121-21.tys.bellsouth.net) |
| 17:20.19 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 17:21.32 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 17:25.18 | *** join/#asterisk kasanop1 (~kasanop@128-68-127-68.broadband.corbina.ru) |
| 17:26.45 | MLNoah | I have an Asterisk 11.5.1 server using res_odbc to provide a phone interface to our timeclock system, and i've extended it with res_xmpp to allow staff to use our internal Jabber server to clock in/out, review time cards, etc. in the process of a user interacting with the XMPP interface, Asterisk challenges for a PIN and waits for a response with JABBER_RECEIVE(). However, if another user... |
| 17:26.45 | MLNoah | ...sends an XMPP message while Asterisk is waiting for this reply, the incoming new message "forcibly times out" the JABBER_RECEIVE() function. |
| 17:27.50 | MLNoah | is there some way I can handle the "bump out" from JABBER_RECEIVE() and keep waiting for the duration of the initial timeout? should I report this as a bug? |
| 17:30.37 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:33.52 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 17:34.46 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 17:35.16 | *** join/#asterisk yaago (~kresp0@gateway/tor-sasl/kresp0) |
| 17:39.00 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:39.53 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:40.28 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:40.42 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:51.49 | cusco | outtolunc: so you confirm that the TE / NT reffers to the timing source (in our digium card) ? Telco states that we must configure our card as "Network Termination" |
| 17:51.56 | cusco | thus as NT |
| 17:52.12 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 17:52.17 | cusco | and they further state that is the mode of how the pbx comunicates with the network |
| 17:53.30 | outtolunc | cusco: that would imply they expect timing from you. what board do you have, where you located, goals for use? |
| 17:55.03 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 17:55.41 | outtolunc | As an example.. if i were building on of my usual systems (multiple 8 port sangomas A108) in the US for outbound.. i would want them to provide timing for the 10+ NI2 PRI's i would request.. but lets say i am in a verizon area.. i'd have to change the NI2 to DMS100 |
| 17:57.04 | cusco | outtolunc: are you sure? |
| 17:57.10 | cusco | I fiddled with timming sources before |
| 17:57.36 | cusco | right now we're testing the line on a TE120P |
| 17:58.24 | cusco | the only time I fiddled with timing sources, was on a 4 span card, usign 2 different telco's |
| 17:58.43 | cusco | and a single card could only source timing from one telco |
| 17:59.12 | cusco | so in system.conf (dahdi) right now I have: |
| 17:59.14 | cusco | span=1,1,0,ccs,hdb3,crc4 |
| 17:59.37 | cusco | but telc sstates its not related to this 'timing source' |
| 17:59.47 | outtolunc | note: multi carrier circuit setups.. yes you should provide the timing |
| 18:00.19 | outtolunc | otherwise the card(s) will do timing resyncs and cause issues with the other carrier circuits |
| 18:00.36 | cusco | also config example states 'pstn will never be a slave to you' |
| 18:01.16 | cusco | so in system.conf |
| 18:01.26 | cusco | I should change "span=1,1,0,ccs,hdb3,crc4" to "span=1,0,0,ccs,hdb3,crc4" |
| 18:01.27 | cusco | ? |
| 18:01.41 | outtolunc | that 1,1 (second 1) implies you are the timing source |
| 18:01.49 | outtolunc | (sorry dealing with phone calls also) |
| 18:01.55 | cusco | no problem |
| 18:02.16 | cusco | the 1 implies that I AM timing source? ouch, ok |
| 18:02.25 | cusco | I had that wrong |
| 18:02.28 | outtolunc | yes |
| 18:04.48 | outtolunc | Note: the debug/logs should report if you have it set to TE and they think they are... |
| 18:05.11 | outtolunc | iirc, 'they think they are CPE, when we are'.. |
| 18:06.51 | outtolunc | Also note: some older cards didn't truly change timing until the module was unloaded/reloaded |
| 18:10.19 | outtolunc | remember to set your chan_dahdi.conf also signalling pri_net (vice pri_cpe) |
| 18:11.23 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 18:24.52 | outtolunc | 426 Upgrade Required |
| 18:24.57 | outtolunc | grr oops |
| 18:28.14 | *** join/#asterisk tris (~tristan@2001:1868:a00a::4) |
| 18:37.44 | *** join/#asterisk liamjfoy (~liamjfoy@unaffiliated/liamjfoy) |
| 18:39.38 | liamjfoy | Hey, does anyone have any experience with Ciscos 7960 & Asterisk 10.12.1? |
| 18:39.41 | *** join/#asterisk kasanop1 (~kasanop@85.94.32.98) |
| 18:39.43 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 18:39.46 | [TK]D-Fender | liamjfoy: That isn't the latest from that branch, and it is basically dead as it is... |
| 18:39.47 | [TK]D-Fender | liamjfoy: What is your question concerning them? |
| 18:39.49 | liamjfoy | [TK]D-Fender: Ah ok, worth upgrading then, yes? |
| 18:39.49 | *** join/#asterisk dongola7_ (~dongola7@unaffiliated/blair/x-0911782) |
| 18:39.50 | liamjfoy | [TK]D-Fender: I can't get the device to register, the Cisco 303 devices work perfectly well |
| 18:39.58 | liamjfoy | Always returning 401s, which seems quite well documented via Google - but I can't get them to register |
| 18:40.08 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 18:40.11 | *** join/#asterisk Sophira (~sophie@neo.theblob.org) |
| 18:40.51 | [TK]D-Fender | liamjfoy: should be "nat=never" IIRC |
| 18:41.19 | liamjfoy | I |
| 18:41.41 | liamjfoy | I'm going to try that tomorrow, no access right now... was just seeing if anyone else had this issue =) |
| 18:42.10 | *** join/#asterisk nny (~Scott@cpe-066-057-212-252.sc.res.rr.com) |
| 18:42.32 | nny | does anyone have a non six hoop jump through and sacrifice animals way to just record a message in windows and convert to an asterisk friendly format? |
| 18:43.00 | nny | The digium online converter is busted, windows records in it's own WMA format, conversion to WAV still won't work with sox and I am gonna choke a puppy |
| 18:44.30 | [TK]D-Fender | Windows records with the settings you tell it to |
| 18:44.35 | [TK]D-Fender | You aren't forced to use WMA |
| 18:45.03 | nny | [TK]D-Fender: hmm which program? I am using sound recorder atm |
| 18:45.06 | [TK]D-Fender | And forget Windows' tools... use a nive audio program like Audacity, etc |
| 18:47.11 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 18:47.16 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 18:48.15 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 18:48.54 | [TK]D-Fender | Hrm.. seems the new app IS crap |
| 18:49.01 | [TK]D-Fender | old one used to give options... |
| 18:49.03 | *** join/#asterisk jeff_oncell (~jeffasing@67.219.32.137) |
| 18:49.04 | [TK]D-Fender | Just install Audacity |
| 18:49.17 | [TK]D-Fender | Makes your life a lot better |
| 18:49.25 | nny | [TK]D-Fender: yeah I haven't used Audacity to record (I do use it for other things and have it installed) so I'll just find a tutorial and set it up. |
| 18:50.15 | nny | assuming 16 bit PCM mono is best for Audacity, then drop to 8 with sox? |
| 18:50.19 | [TK]D-Fender | nny: New Project. Select Mic to record. Click on VU meter to see it there and adjust gains. click record. Done |
| 18:50.26 | nny | er nm |
| 18:50.28 | nny | it has 8 |
| 18:50.35 | [TK]D-Fender | 16bit 8 khz. |
| 18:51.03 | nny | [TK]D-Fender: this should work thanks. Umm... do you want this puppy? It's kind of limp |
| 18:51.07 | [TK]D-Fender | Don't mix up the bit-depth with the sample rate :) |
| 18:53.53 | nny | [TK]D-Fender: one question, why does hitting record create a new audio track in Audacity? |
| 18:54.40 | [TK]D-Fender | You have to choose if you want to overdub... |
| 18:54.51 | [TK]D-Fender | It always records new layers otherwise.. |
| 18:55.41 | jeff_oncell | Where is the best place to find old asterisk documentation? |
| 18:55.41 | nny | k |
| 18:57.22 | cusco | outtolunc: no I never get to such logs |
| 18:57.25 | cusco | outtolunc: I only got there when I plugged a loop in the pri card |
| 18:57.35 | cusco | outtolunc: I never get comunication with PRI :( |
| 18:57.47 | cusco | basically all I see in pri debug is sending SABME |
| 18:57.58 | [TK]D-Fender | jeff_oncell: How old? |
| 18:58.11 | [TK]D-Fender | ~wikis |
| 18:58.11 | infobot | methinks wikis is VoIP Wiki covering FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners. For Asterisk, see ~asteriskwiki |
| 18:58.16 | [TK]D-Fender | ^ mostly dated stuf... |
| 18:58.23 | jeff_oncell | I need 1.4 |
| 18:58.39 | [TK]D-Fender | ther is the v2 of the BOOK that covers that more specifically. |
| 18:58.45 | [TK]D-Fender | Got something in particular? |
| 18:59.04 | outtolunc | cusco: SABME simply means the layer2 -> layer3 acceptance hasn't occurred (is the circuit 'turned-up' (continuity <> 'turned-up')) |
| 18:59.28 | jeff_oncell | I'm just guessing I'm going to have lots, and lots of questions over the next week or two |
| 18:59.33 | cusco | yes thats the problem |
| 18:59.49 | cusco | outtolunc: I insert the loop in their equipment and they got link up signal |
| 18:59.55 | cusco | if I connect it to our pri card |
| 19:00.01 | cusco | nor me nor they, get signal up |
| 19:01.17 | outtolunc | cusco: did you make sure to reload the modules between changes? |
| 19:01.38 | cusco | what changes, in the timing source? |
| 19:02.01 | outtolunc | cusco: yes to system.conf and chan_dahdi.conf |
| 19:02.16 | cusco | meanwhile telco has the line turned off, so I'm waiting for them to turn it on... I gues sit will only happen on monday |
| 19:02.20 | cusco | but this was before that change |
| 19:02.25 | cusco | I should see such error |
| 19:02.33 | cusco | 'we're cpe but they're think they're cpe too' |
| 19:02.34 | cusco | or something |
| 19:02.44 | cusco | but we never got that far |
| 19:02.56 | outtolunc | cusco: if they 'turned it off' then yes.. you'll see SABME's |
| 19:03.14 | cusco | yes but what I'm stating is all I've seen is SABME's |
| 19:03.21 | cusco | with this telco |
| 19:03.53 | outtolunc | cusco: just make a t1 loopback plug/cable and is if the port(s) steady (no errors) |
| 19:04.05 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 19:04.12 | cusco | I did just that. seems that you're not reading me |
| 19:04.25 | cusco | and thats when I got the 'cpe' debug/error message |
| 19:04.43 | outtolunc | cusco: being a 2 port card, you can set 1 as NT and 1 as TE (system and dahdi confs) |
| 19:04.58 | cusco | its a 1 port card (also stated the model above) |
| 19:05.14 | cusco | but... |
| 19:05.23 | cusco | I've tested it today with another telco |
| 19:05.32 | cusco | and stated up, active |
| 19:05.43 | cusco | so.... I'm betting on a telco's problem |
| 19:06.13 | outtolunc | cusco: its a telco or telco config problem (assuming the cabling to the NIU's are the same) |
| 19:06.39 | cusco | I've tried different cables, tried crossed over cable |
| 19:06.53 | cusco | (we had some telco that required crossed over cable) |
| 19:09.18 | *** part/#asterisk nny (~Scott@cpe-066-057-212-252.sc.res.rr.com) |
| 19:26.37 | jeff_oncell | I'm trying to setup SIP, and getting a 404 every time I try to register with my server, any ideas where to start? |
| 19:43.43 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:43.43 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.6.0 (2013/10/21), 10.12.3 (2013/08/27), 1.8.24.0 (2013/10/21), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 19:48.21 | ChannelZ-Wk | You only need 3 or so |
| 19:50.11 | ChannelZ-Wk | turn on sip debug (sip set debug on) and try, pastebin the results |
| 19:51.33 | jeff_oncell | Ahhh, well that got some output |
| 19:51.53 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 19:53.13 | jeff_oncell | http://pastebin.com/T2cQJHKd |
| 19:54.58 | *** join/#asterisk mitchrodrigues (~mitchrodr@204-16-155-82-static.ipnetworksinc.net) |
| 19:58.32 | *** part/#asterisk esaym153 (~esaym153@216-45-91-132.gvec.net) |
| 20:12.37 | ChannelZ-Wk | sorry been off doing other thigns |
| 20:13.12 | ChannelZ-Wk | Do yo have a peer named [jeff] in sip.conf? |
| 20:13.12 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:16.50 | *** join/#asterisk tzafrir (~tzafrir@37.26.147.204) |
| 20:17.45 | jeff_oncell | No, I had a peer named vagrant that I was trying to match, with host set to 10.91.0.1 |
| 20:21.38 | ChannelZ-Wk | Is it loaded? sip show peers |
| 20:21.55 | ChannelZ-Wk | and/or pb your sip.conf (just XXX your secrets) |
| 20:22.42 | jeff_oncell | vagrant 10.91.0.1 N 5060 Unmonitored |
| 20:23.01 | jeff_oncell | Relevant line from sip show peers |
| 20:24.00 | ChannelZ-Wk | So rename the user you typed into zoiper to vagrant |
| 20:26.23 | jeff_oncell | Okay, I'm using x-lite, because zoiper was doing weird things with my system level audio, but I did that there, and it doesn't seem to make any difference |
| 20:26.45 | ChannelZ-Wk | sorry yes, my brain substituted zoiper |
| 20:28.47 | jeff_oncell | Okay, I'm actually getting a 403 when I try to register now |
| 20:32.28 | ChannelZ-Wk | Make host=dynamic, or you don't need to register |
| 20:32.56 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 20:34.18 | jeff_oncell | Okay |
| 20:34.40 | jeff_oncell | That got me to an error inside the application I'm trying to test, which is great |
| 20:35.11 | ChannelZ-Wk | The.. dialplan application? |
| 20:35.48 | jeff_oncell | Yes |
| 20:36.14 | jeff_oncell | Well, it was an error in a php script called over AGI |
| 20:36.25 | ChannelZ-Wk | ah |
| 20:37.17 | jeff_oncell | so at some point I'll need a way to correctly setup the SIP.conf, but this works for me for now. I really appreciate all your help |
| 20:37.42 | ChannelZ-Wk | Well, I think you just did if things are making it into your dialplan. |
| 20:38.29 | jeff_oncell | Yeah, though, eventually I need a configuration that works both on my machine and in production |
| 20:41.11 | ChannelZ-Wk | Not that much of a difference :) |
| 20:56.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.62) |
| 20:57.15 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 20:58.13 | *** join/#asterisk micols (~t@shell1.rlogin.dk) |
| 21:16.15 | *** join/#asterisk tty-1 (~jumperswi@ltea-047-067-204-221.pools.arcor-ip.net) |
| 21:16.17 | tty-1 | hi |
| 21:16.20 | tty-1 | ~did |
| 21:16.20 | infobot | i heard did is Direct Inward Dialing, or just a phone number |
| 21:17.00 | tty-1 | ~tpsp |
| 21:17.12 | tty-1 | ~pts |
| 21:17.12 | infobot | hmm... pts is Public Test Shard for Rift. For more information on how to get started see this post: http://bit.ly/kgztys |
| 21:17.17 | tty-1 | ~ptsp |
| 21:17.27 | Penguin | itsp |
| 21:17.44 | Penguin | ptsn |
| 21:18.05 | tty-1 | hi Penguin! |
| 21:18.08 | tty-1 | ~itsp |
| 21:18.08 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:18.10 | tty-1 | ~ptsn |
| 21:18.26 | Penguin | I mistyped it.. pstn |
| 21:18.35 | tty-1 | ~pstn |
| 21:18.35 | infobot | methinks pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 21:18.47 | tty-1 | ah |
| 21:18.51 | tty-1 | I will pin those down |
| 21:18.54 | tty-1 | Very important to know |
| 21:19.06 | Penguin | I was on the phone, typing with one hand and messed up. |
| 21:20.06 | tty-1 | Penguin: yes, this happens easily then |
| 21:20.37 | tty-1 | Penguin: how can I check on asterisk cli if asterisk is listening for incoming calls / successfully logged in to sip provider? |
| 21:20.43 | tty-1 | *on sip provider |
| 21:21.19 | Penguin | You can't know definitively by looking at that info. |
| 21:21.28 | Penguin | But you can check your registry: sip show registry |
| 21:21.37 | Penguin | This will show if your asterisk is registered to the provider. In most cases, if you are registered, calls will be sent to you. |
| 21:21.54 | Penguin | But that doesn't mean your asterisk is configured to handle the calls that come in. |
| 21:22.09 | tty-1 | ok |
| 21:22.20 | tty-1 | I like how fast asterisk starts up and generally is. |
| 21:23.54 | cusco | hi |
| 21:24.15 | tty-1 | cusco: hi |
| 21:24.25 | tty-1 | Penguin: yes, it is registered |
| 21:24.32 | cusco | sup |
| 21:24.46 | cusco | ~linkedid |
| 21:24.52 | cusco | ow |
| 21:24.53 | cusco | :( |
| 21:25.01 | Penguin | linke did? |
| 21:25.06 | Penguin | linked id? |
| 21:25.12 | cusco | ~test |
| 21:25.13 | infobot | it has been said that test is not funny |
| 21:25.19 | cusco | ay! |
| 21:25.20 | cusco | :) |
| 21:25.30 | cusco | we're on 1.6 and have been for too long |
| 21:25.48 | tty-1 | Penguin: oh this is good. when in cli I get a feedback when calls goes in |
| 21:25.49 | cusco | we really need to move to 1.8 or upwards (trying latest .11 ) |
| 21:25.53 | cusco | to use linkedid |
| 21:25.58 | cusco | I'm worried |
| 21:26.04 | cusco | queue_log format changes |
| 21:26.12 | Penguin | lin kedid? |
| 21:26.12 | Penguin | What did you really mean? |
| 21:26.12 | Penguin | ~asterisk versions |
| 21:26.12 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 21:26.15 | cusco | we created a view in mysql to mirror old behaviour lol |
| 21:26.20 | cusco | linked id |
| 21:26.22 | Penguin | ~versions |
| 21:26.23 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 21:26.23 | Penguin | hmm |
| 21:26.29 | cusco | as a variable |
| 21:26.45 | Penguin | ~astversions |
| 21:26.49 | Penguin | Oh, lag. |
| 21:27.02 | cusco | I'm aware 1.6 is deprecated |
| 21:27.49 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 21:30.55 | tty-1 | Penguin: there is the dial (command ?) in /etc/asterisk/extensions.conf which second parameter should some kind of hostname, but I get this in the cli: chan_sip.c:5470 create_addr: No such host |
| 21:31.23 | tty-1 | Penguin: What kind of host? |
| 21:31.58 | Penguin | tty-1: The Dial() application syntax is as follows: Tech/device[/extension],timeout,options |
| 21:32.13 | Penguin | or Tech/extension@host,timeout,options |
| 21:32.21 | Penguin | ~book |
| 21:32.21 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:32.34 | tty-1 | Penguin: OK, so just Dial(SIP/<what device?>) |
| 21:32.47 | Penguin | The devices are configured in sip.conf. |
| 21:32.50 | tty-1 | Penguin: ah, it wants to know what device can dial, right? |
| 21:34.49 | Penguin | It's all in the book. |
| 21:35.05 | tty-1 | ok |
| 21:35.33 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 21:40.08 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 21:42.30 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 21:42.40 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 21:44.11 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 21:54.04 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:56.55 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 21:58.16 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.59) |
| 22:01.22 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 22:03.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.68) |
| 22:03.30 | *** join/#asterisk mmlj4 (1000@ip98-163-253-141.no.no.cox.net) |
| 22:13.57 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 22:15.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:18.20 | *** join/#asterisk smirker (~x@CPE-144-137-156-146.lnse3.cha.bigpond.net.au) |
| 22:53.53 | *** join/#asterisk rogersja (~rogersja@207.164.2.174) |
| 22:55.45 | rogersja | any really good recommendations for a sip soft phone for iOS that supports H264, or H263 or H263p. other than zoiper or bria |
| 23:02.47 | *** join/#asterisk lvlinux (~n1gg@adsl-98-86-121-21.tys.bellsouth.net) |
| 23:06.14 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 23:18.45 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 23:26.17 | *** join/#asterisk rdegges (rdegges@gateway/shell/ircrelay.com/x-sdfcepwdbwyonqcc) |
| 23:29.14 | tty-1 | exten => 0123456789,1,Dial(SIP/arcor,10) |
| 23:29.29 | tty-1 | exten => 0123456789,2,VoiceMail(primary,s|u) |
| 23:29.35 | tty-1 | Why doesn't it wait for 10 seconds. |
| 23:30.45 | ChannelZ-Wk | If the dial failed... |
| 23:31.23 | ChannelZ-Wk | IE the peer is offline, non-existant, or otherwise rejected the call.. |
| 23:32.05 | ChannelZ-Wk | Presumably if the peer was online and accepting calls it would ring for 10 seconds. |
| 23:34.54 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 23:37.00 | tty-1 | ChannelZ-Wk: yes, some other peer is online |
| 23:37.16 | tty-1 | ChannelZ-Wk: so a Wait-command has to be inserted between Dial and VoiceMail, right? |
| 23:37.43 | WIMPy | That's probably not what you want. |
| 23:38.02 | ChannelZ-Wk | I don't know what you want now :) |
| 23:38.49 | ChannelZ-Wk | If your device isn't rining for at least 10 seconds before dumping you to voicemail, then the call failed for some reason. Look at the console. |
| 23:39.42 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 23:42.18 | *** join/#asterisk jsgoecke (~jsgoecke@c-50-161-77-217.hsd1.ca.comcast.net) |
| 23:49.14 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |