| 00:00.39 | watchy | qwell: i can tell its a poly by mac, but i wanna tell what model it is |
| 00:01.04 | watchy | qwell: ive seen how switchvox can detect new phones on teh network and i'd like to beable to do that myself |
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| 00:16.54 | drfreeze | Can someone take a look at the call parking problem |
| 00:16.55 | drfreeze | http://gist.github.com/174298 |
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| 00:25.25 | drfreeze | Ok, I'm kind of figureing out this ParkAndAnnounce and announce call, but don't know how to get it to speak back the digits |
| 00:25.54 | drfreeze | Passing the argument Sip/num calls the phone doing the parking |
| 00:26.11 | drfreeze | How do I just get it to say the parking location? |
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| 00:27.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 00:34.55 | SwK | anyone around HSV have an extra TDM400 or 2? |
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| 00:49.43 | kb3ien | http://pastebin.com/m49863655 not getting this regex thing quite right... seems to be in my set()... any suggestions? |
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| 00:55.13 | JafoJ | Anyone know PRI debugging here? |
| 00:55.21 | Carlos_PHX | A few people. |
| 00:55.25 | JafoJ | If you turn PRI debugging on and have verbosity set to 0, and a PRI error happens, is the PRI error directly coinciding with the PRI debug at that same time. Or is one independant of the other in Asterisk? |
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| 01:11.49 | rob0 | SwK, mine was stolen in April :( |
| 01:12.16 | rob0 | I do have one FXO module that wasn't in it. |
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| 01:14.08 | dlynes | swears at Freepbx. |
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| 01:20.10 | voipmonk | :) |
| 01:22.57 | dlynes | Default timeouts for sip calls are 20s? That's crazy |
| 01:23.29 | dlynes | And maximum timeout of 60s means 4 in every 5 international long distance calls fail |
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| 02:43.33 | manxpower | ~answers |
| 02:43.34 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6: /path/to/src/asterisk/UPGRADE*.txt |
| 02:43.40 | manxpower | LONG LIVE INFOBOT! |
| 02:48.44 | shido6 | ~botsnack |
| 02:48.44 | infobot | shido6: :) |
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| 03:20.24 | kb3ien | im calling dial(local/NNNN@cntx); what do i do in cntx to return a result code? just call Congestion() etc, as normal? |
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| 03:25.51 | manxpower | kb3ien: have you confirmed that the DIALRESULT and HANGUPCAUSE of the child dial is not passed up to the Local/ dial? |
| 03:26.42 | manxpower | if you've confirmed it does not, look at the docs for Hangup, I think you can pass it a hangupcause |
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| 03:46.30 | manxpower | For any that find it useful: http://www.fnords.org/~eric/polycom-config-examples/ |
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| 04:56.52 | AussieGuy | do linksys IP phones need a power over ethernet switch or will a regular switch work? |
| 04:58.30 | [TK]D-Fender | AussieGuy: Regular switch won't power them |
| 04:59.27 | AussieGuy | POE switch then? |
| 05:00.17 | [TK]D-Fender | AussieGuy: If thats what you want to do, sure |
| 05:00.55 | AussieGuy | or would they take power from their own power supply |
| 05:02.20 | [TK]D-Fender | AussieGuy: thats the point |
| 05:02.31 | [TK]D-Fender | AussieGuy: Keep in mind if the PS is a separate purchase |
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| 05:18.38 | denon | actually, dont most linksys phones ship with a PS? |
| 05:18.43 | denon | but yeah, that's definitely not the norm |
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| 05:25.13 | drmessano | SPA-942 and SPA-922 do not |
| 05:25.41 | denon | ah ok, been a while since I laid hands on a new linksys phone personally |
| 05:26.26 | drmessano | They're the PoE variants, so ++PoE --Power Supply.. Guess you can't have cake+eat too |
| 05:27.27 | denon | well, it's only logical |
| 05:27.35 | denon | got enough spare power supplies laying around as it is |
| 05:27.55 | denon | I'd rather not pay for more junk I don't need (then again, they probably dont charge extra, just take extra profit) |
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| 05:29.40 | drmessano | The opportunity to use the PS with the built-in switch to be able to power the phone and use the passthru for the workstation is there, but you need PoE or to buy a seperate PS to do it, which is sucky, and somewhat odd.. as I would think if theres any situation you would want that, it would be for the lower end phones |
| 05:30.12 | drmessano | If youre too cheap for PoE, springing another $20 to save yourself from pulling another drop would be nice |
| 05:33.09 | drmessano | Sorry, forgot we're talking about Cisco here |
| 05:33.12 | drmessano | moves along |
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| 05:43.35 | shadowalke | Good evening all I am looking for an example of how to use mix monitor in a dialplan |
| 05:44.11 | shadowalke | using a Set to establish the file name structure |
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| 06:30.13 | tokozedg | hello, anyone can advise server hardware configuration for asterisk about 800 users? |
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| 06:37.37 | alunca | 800 users? damn hehe I don't know |
| 06:38.51 | tokozedg | alunca, can asterisk work with 800 users? |
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| 06:40.32 | alunca | tokozedg I really don't know if asterisk can handle that much users |
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| 06:42.25 | tokozedg | as i know asterisk also cant make a good clustering? |
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| 06:44.40 | fcois93 | hello all |
| 06:44.46 | J4k3 | 800 users is not that bad |
| 06:44.51 | J4k3 | as long as they're not all on the phone at once |
| 06:44.59 | J4k3 | asterisk doesn't gracefully scale like that |
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| 06:47.41 | tokozedg | J4k3, most of them will be online |
| 06:48.25 | tokozedg | the maximum simultaneous call will be about 250 |
| 06:48.34 | alunca | tokozedg how many will be online at the same time? |
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| 06:51.08 | tokozedg | alunca, there will be hardware phones, and turned on almost always |
| 06:52.38 | kaldemar | tokozedg: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 06:52.58 | tokozedg | alunca, u know asterisk application konversation, someone wrote that he tested it with 1000 calls, 1000 users were in conference |
| 06:54.16 | tokozedg | kaldemar, nice tut |
| 06:57.13 | alunca | how can I have a license to use g792a on an ATA? |
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| 07:01.22 | maour | how can i compile asterisk in release mode ? |
| 07:02.28 | maour | i think when i compile it using make it would compiled in debug mode , right ? |
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| 07:03.49 | kaldemar | maour: not by default |
| 07:04.19 | maour | kaldemar: you mean by default it's on release mode ? |
| 07:04.29 | kaldemar | there is no release mode |
| 07:04.52 | kaldemar | you'll get some compiler flags in "make menuselect". look at those. |
| 07:05.04 | maour | ok |
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| 07:14.09 | zeeesh | can i run freeswitch and asterisk at the same time at the same machine ? |
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| 07:21.11 | fiddur | zeeesh: With the correct settings, you should be able to run several asterisk on the same machine at least... ..though I've never tried it. |
| 07:22.59 | fiddur | zeeesh: Just make sure they dont use the same ports, or the same temporary files.. |
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| 07:51.55 | ttl_work | I have a asterisk 1.6 server and 3 sip DID numbers configured on it from the same SIP provider, how do I route incoming calls for each DID to separate internal sip phones? |
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| 08:19.38 | alunca | ttl_work imbound routes, point it to the extensions of the DID (from Trunk) |
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| 08:30.50 | ttl_work | alunca: ok, thanks! |
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| 08:33.51 | fcois93 | I try to install a card the TC400P, I follow the manual TC400B-user-manual-1.pdf. at the chapter 3 it tell me to insert mode=g729to be able to have g729 at CLI>show transcoder, but I don't know where!know you? |
| 08:46.07 | dandre | hello, |
| 08:46.40 | dandre | is there a good opensource tsp (tapi) driver for asterisk? |
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| 09:57.38 | garymc | join #ubuntu-server |
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| 11:26.56 | Sniper_voip | Hi all, is there a specific CLI command in asterisk that fetch if a specific username is in call or not? |
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| 11:33.33 | kaldemar | Sniper_voip: what do you mean by a username? |
| 11:33.57 | Sniper_voip | kaldemar, the UserID used for regitration |
| 11:34.14 | fiddur | should we assume sip? |
| 11:34.21 | kaldemar | for what? do you mean a SIP device? |
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| 11:36.40 | c4rg | anyone knowing inner workings of asterisk? |
| 11:37.18 | Chainsaw | c4rg: Several people here, yes. You can skip the intro and ask the important questions :) |
| 11:37.42 | kaldemar | Sniper_voip: core show channels [concise] will show you ongoing calls. you'll get the info by parsing that. |
| 11:38.27 | Sniper_voip | kaldemar, yes but I do not need to see all calls...I just need to see if a specific userID is involved in a call (Like making grep) |
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| 11:40.59 | c4rg | ok, so suppose I have two IAX clients - A and B. A calls B via asterisk. When A sends an RTP packet to asterisk - when asterisk will deliver this data/packet to B? (alaw is the codec at every leg) |
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| 11:54.25 | kaldemar | Sniper_voip: then grep it. for example "asterisk -rx 'core show channels concise' | grep SIP/ID" in linux or whatever you run asterisk on. but that's not a good way to use in any third party application. |
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| 12:21.00 | datacompboy | Hi all:) anybody knows how to make dialplan based on diversion headers? |
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| 12:23.15 | [TK]D-Fender | datacompboy: Meaning? |
| 12:23.31 | [TK]D-Fender | datacompboy: "core show function SIP_HEADER" |
| 12:23.57 | datacompboy | [TK]D-Fender: ooo! thanks. will try to play with it |
| 12:24.43 | *** join/#asterisk coppice (n=chatzill@52.204.17.210.dyn.pacific.net.hk) |
| 12:28.07 | *** join/#asterisk voipmonk (n=voipmonk@74.132.202.71) |
| 12:36.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
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| 12:49.29 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 12:49.36 | garymc | Anyone tell me what Rhino cards are like, as they are half the price |
| 12:49.40 | Skeeter- | gosh, now we dont have any telephone lines and i need to configure with sip provider |
| 12:49.57 | manxpower | ~answers |
| 12:49.58 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6: /path/to/src/asterisk/UPGRADE*.txt |
| 12:50.37 | [TK]D-Fender | manxpower: Pretty comprehensive, is that one your doing? |
| 12:50.57 | leifmadsen | you're* |
| 12:51.04 | [TK]D-Fender | Correct |
| 12:51.08 | leifmadsen | :) |
| 12:51.24 | *** join/#asterisk R0dya (n=R0dya@213-202-142-248.bas502.dsl.esat.net) |
| 12:51.25 | [TK]D-Fender | awards leifmadsen his Grammar Merit Badge |
| 12:51.35 | leifmadsen | gold star! |
| 12:52.17 | garymc | anyone even heard of Rhino cards? |
| 12:52.24 | leifmadsen | I have heard of them, yes. |
| 12:52.53 | coppice | garymc: you make them sound obscure. they aren't |
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| 12:59.24 | [TK]D-Fender | garymc: I don't see them at half the price. |
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| 13:07.56 | garymc | no, the person I got looking for best price on sangoma said they are alot cheaper |
| 13:08.08 | garymc | the rhino equivalent that is |
| 13:08.23 | garymc | as im a beginner am i best going for sangoma? |
| 13:08.25 | ltd | if you want cheap |
| 13:08.30 | ltd | openvox |
| 13:08.37 | garymc | but are they any good? |
| 13:09.04 | garymc | i dont wanna throw £300 on an openvox equivalent to only have to spend £700 on the sangoma later on |
| 13:09.12 | [TK]D-Fender | garymc: Show me a product... |
| 13:09.36 | garymc | which one? |
| 13:09.44 | ltd | if it was for a mission critical application, I would use digium personally |
| 13:09.44 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 13:10.11 | manxpower | garymc: The number one problem with cards other than Digium and Sangoma is that you'll get virtually no support here. |
| 13:10.48 | garymc | ok |
| 13:10.55 | garymc | so you all use digium or sangoma |
| 13:11.03 | Chainsaw | Correct. Digium here. |
| 13:11.07 | [TK]D-Fender | garymc: the Rhino @ half price. the "I had a friend look". considering how little you know about these products I'm betting he has NOT clue what he is comparing |
| 13:11.24 | *** join/#asterisk shadowalke (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 13:11.37 | garymc | lol probably. well he said alot cheaper... chinese whispers an all that :) |
| 13:11.50 | shadowalke | Good morning do you know if you can configure followme to run out of mysql |
| 13:11.51 | shadowalke | ? |
| 13:11.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:12.01 | *** join/#asterisk knobo (n=user@216-55-160.100710.adsl.tele2.no) |
| 13:12.31 | voipmonk | go home |
| 13:12.33 | voipmonk | :) |
| 13:12.38 | knobo | me? |
| 13:12.44 | voipmonk | shadowalke: :) |
| 13:12.50 | Katty | :> |
| 13:12.53 | garymc | equvilent openvox £411 |
| 13:12.54 | Katty | it's another beautiful day! |
| 13:13.16 | manxpower | Not many people here use Xorcom devices either, but they are very active in their driver development. |
| 13:13.19 | [TK]D-Fender | shadowalke: Probably not, and its a worthless app anyways, just code your own. |
| 13:13.41 | [TK]D-Fender | [09:12]<garymc>equvilent openvox £411 <- Link please |
| 13:13.50 | manxpower | Unlike Sangoma, the Xorcom drivers are actually part of Zaptel |
| 13:14.11 | garymc | £481 with vat |
| 13:14.12 | knobo | what happens if zapata.conf says group= (and then nothing)? |
| 13:14.13 | garymc | http://www.voipon.co.uk/openvox-de115e-pci-isdn-pri-card-with-hw-echo-cancellation-p-1707.html |
| 13:14.22 | knobo | will it affect anything? |
| 13:14.26 | manxpower | I *personally* would never use a USB channel bank (Xorcom is a USB channelbank), but other people do. |
| 13:14.57 | shadowalke | voipmonk |
| 13:14.58 | shadowalke | lol |
| 13:14.59 | shadowalke | hey\ |
| 13:15.22 | voipmonk | hello there |
| 13:15.36 | [TK]D-Fender | There are very few places I would recommend a Xorcom style solution, but they have come a log wy in features & price. they have their place. |
| 13:15.47 | kaldemar | knobo: it will prevent channel definitions below it from belonging to another group defined above it. |
| 13:15.47 | coppice | the timing in those xorcoms used to be very troublesome, but they seem to have sorted them out now. I have a 16 port FXS one here that will send FAXes indefinitely without error |
| 13:16.07 | [TK]D-Fender | knobo: Yes, it will blank out the rgoup # for whatever channel declaration follows it |
| 13:16.20 | kaldemar | knobo: that's how it used to behave. i recommend you test it yourself though. |
| 13:16.33 | *** join/#asterisk rossand (n=rossand@bas1-ottawa11-1176120661.dsl.bell.ca) |
| 13:16.40 | manxpower | [TK]D-Fender: I'm impressed with them simply for the fact they have been around for a while. I never thought a USB channelbank would ever work, but obviously they do. |
| 13:16.45 | *** join/#asterisk oberon (n=oberon@89-138-209-249.bb.netvision.net.il) |
| 13:16.59 | oberon | what ports does RTP use ? |
| 13:17.18 | coppice | manxpower: do you realise that USB was designed to carry E1s? |
| 13:17.27 | manxpower | oberon: any ports the two devices agree on. Set the ports in ASTERISK by editing rtp.con |
| 13:17.38 | [TK]D-Fender | manxpower: they can now handle server failover (dual USB), and a few other nifty things. On the high-port scale they are not far off from the price of SIP Gateways per-port. |
| 13:17.56 | Katty | http://www.youtube.com/watch?v=mMHSKXN6qNk <- eppigy |
| 13:18.09 | manxpower | coppice: I did not know that. |
| 13:18.19 | shadowalke | Ihey voipmink |
| 13:18.24 | shadowalke | check your other window |
| 13:18.29 | Katty | eppigy: you will be creeped out, but it will get stuck in your head :< |
| 13:18.32 | manxpower | I can tell you that I've had NOTHING but TROUBLE trying to do any kind of high speed data on USB on linux. |
| 13:18.48 | coppice | manxpower: if you look at the original USB specs, some of them are from people at nortel defining ISDN over USB, and such like |
| 13:19.13 | coppice | manxpower: and digium E1 cards work perfectly? |
| 13:19.21 | manxpower | I actually had to switch to FreeBSD just to get a Verizon EVDO device to work at anything close to the speed it works under Windows. |
| 13:19.49 | manxpower | I'll stick to PCI for telecom stuff, thankyouverymuch |
| 13:20.30 | coppice | but the PCI stuff looses data, too, as soon as there is too much load. you're trying to do real time stuff in a far from real time environment |
| 13:20.57 | ocnarf | Hi, need help on my tce400 card. I dont know where to set mode = g729. I will be using 120 channels for g729 |
| 13:21.37 | manxpower | ocnarf: I believe you set it exactly the way you would set it if you were using software G729 instead of hardware G729. |
| 13:21.43 | [TK]D-Fender | ocnarf: sip.conf, iax.conf.... |
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| 13:22.34 | ocnarf | i read the manual and it says i need to change mode = g729 if i will be using 120 channels. |
| 13:22.43 | Katty | today is brought to you by the letter B! the number 2! and the triangle! |
| 13:22.46 | ocnarf | let me try that, ill let you know |
| 13:22.48 | ocnarf | thanks |
| 13:23.04 | eppigy | Katty: :D |
| 13:23.15 | Katty | eppigy: good morning sunshine! |
| 13:23.34 | Katty | hugs eppigy |
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| 13:26.04 | knobo | thanx |
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| 13:29.21 | ocnarf | manxpower: its still says that 0/0 encoders/decoders of 92 channels are in use. |
| 13:29.51 | ocnarf | the manual says that if i set it to mode =g729, 120 channels would be used |
| 13:31.03 | radovan | hi guys |
| 13:32.28 | radovan | when i use hw encoder decoder like tc400b, when it's using all channels, then the transcoding goes sw? |
| 13:33.23 | radovan | i mean transcoding of more channels |
| 13:34.52 | eppigy | falls asleep in Katty's lap |
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| 14:08.06 | Katty | ATTENTION |
| 14:08.14 | Katty | IT IS EPPIGY"S BIRFDAY |
| 14:08.22 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733667.dsl.bell.ca) |
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| 14:12.52 | Katty | :< |
| 14:13.01 | Katty | no one's gonna say happy birfday to eppigy? )= |
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| 14:15.05 | _trine | is there anyone here who can help me with the correct syntax for genmc? |
| 14:16.07 | manxpower | genmc? |
| 14:16.59 | _trine | yes genmc is a linksys application used to generate mini certificates |
| 14:17.08 | _trine | it's a cli app |
| 14:17.19 | radovan | is it working? |
| 14:17.22 | manxpower | _trine: I wish you the best of luck |
| 14:17.43 | radovan | i tried it, and it was scrambling voice |
| 14:18.22 | coppice | people pay a lot for voice scramblers |
| 14:18.31 | _trine | well I have got some mini certificates working fine I got from voxilla but I would like to make my own |
| 14:18.45 | manxpower | coppice: I thought they only payed a lot for voice DEscramblers? |
| 14:19.29 | coppice | they tend to come it pairs |
| 14:19.48 | beek | What's so special about a voice scrambler. An old 10BaseT Hub, some cat 3 wiring and voila: voice scrambling! |
| 14:20.38 | beek | How about VoARCNET? |
| 14:23.55 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:24.29 | Naikrovek | say a user's extension has changed; how do you automatically transfer someone that dials their old extension to the new extension |
| 14:25.00 | Carlos_PHX | Add another dialplan entry for the old one. |
| 14:25.15 | Carlos_PHX | A user can have unlimited dialplans to reach them. |
| 14:25.50 | manxpower | exten => oldexten,1,Goto(newexten1) |
| 14:25.53 | manxpower | ..er. |
| 14:25.58 | kaldemar | +, |
| 14:25.59 | manxpower | exten => oldexten,1,Goto(newexten,1) |
| 14:27.14 | Naikrovek | tyvm |
| 14:27.40 | manxpower | You could not find the info in the Asterisk book? |
| 14:27.57 | Naikrovek | adobe reader broken atm |
| 14:28.00 | Naikrovek | not sure what's up with it |
| 14:28.04 | Naikrovek | some auto update failed it hink |
| 14:28.06 | Naikrovek | i think* |
| 14:29.56 | leifmadsen | Naikrovek: http://astbook.asteriskdocs.org |
| 14:29.59 | leifmadsen | HTML version |
| 14:30.06 | Naikrovek | ah cool |
| 14:30.14 | Naikrovek | i downloaded the pdf and forgot about the site |
| 14:30.17 | Naikrovek | tyvm |
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| 15:07.35 | *** join/#asterisk wopsy (n=80475@AToulouse-754-1-1-6.w90-11.abo.wanadoo.fr) |
| 15:07.42 | wopsy | hi |
| 15:08.09 | wopsy | someone can help me ? i've a problem with asterisk and LDAP . pm pls :) |
| 15:11.03 | leifmadsen | wopsy: why not just ask a question to a room of a lot of people? |
| 15:11.14 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:12.51 | Naikrovek | oh Asterisk... sometimes you're awesome and smoetimes you really f*king suck |
| 15:14.31 | wopsy | I use asterisk 1.6 connected with my LDAP. LDAP users can call a person configured in sip.conf but can't receive call. Also , LDAP user can't call another LDAP user |
| 15:14.45 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 15:14.58 | wopsy | more info and error report > http://forums.digium.com/viewtopic.php?t=70947 |
| 15:15.15 | wopsy | any idea ? |
| 15:16.04 | leifmadsen | wopsy: potentially a known issue: https://issues.asterisk.org/view.php?id=13861 |
| 15:17.38 | [TK]D-Fender | Naikrovek: What now? |
| 15:17.48 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 15:18.11 | Naikrovek | i won't bother anyone in here with my inferior trixbox system, [TK]D-Fender |
| 15:18.50 | [TK]D-Fender | Naikrovek: So basically * is fine and your retarded GUI can "suck it" :p |
| 15:18.57 | Naikrovek | in this case it's asterisk |
| 15:19.10 | Naikrovek | but no one in here will ever believe that so i won't even bring it up |
| 15:19.12 | [TK]D-Fender | Naikrovek: Oh? c'mon share... whats the gripe of the day? |
| 15:19.30 | [TK]D-Fender | Naikrovek: And don't try getting away with a pot-shot shelling like that.. |
| 15:19.41 | Naikrovek | you get away with it all the time |
| 15:19.45 | *** part/#asterisk larin (n=larin@mail.cs-service.by) |
| 15:20.03 | Naikrovek | no, i'm going to figure this one out myself |
| 15:20.07 | Naikrovek | come hell or high water |
| 15:20.22 | Naikrovek | and if i'm wrong, and it does turn out to be a trixbox issue, i'll say so |
| 15:20.23 | [TK]D-Fender | Naikrovek: Do I? Feel free to call me on any specific point. |
| 15:20.44 | wopsy | leifmadsen: this issue doesn't fix my problem . Moreover i don't have AstAccountDefaultUser in my LDAP shema ( i use http://www.voip-info.org/wiki/view/LDAP ) |
| 15:21.14 | leifmadsen | shrugs |
| 15:22.09 | [TK]D-Fender | Naikrovek: And the "I'll figure it out on my own" is a nifty way to leave your claim of *'s fault unchallenged since you aren't backing it up and will likely (as so many others) simply not mention whatever it is that you found. |
| 15:22.20 | Naikrovek | well maybe i'm wrong |
| 15:22.28 | [TK]D-Fender | Naikrovek: But you're welcome to your world-view... |
| 15:22.34 | manxpower | If he's a trixbox user just leave well enough alone, [TK]D-Fender |
| 15:22.36 | Naikrovek | and in being wrong i'll unleash the wrath of a thousand IRC channel participants |
| 15:22.37 | [TK]D-Fender | Naikrovek: Share and we can speed up the process a bit |
| 15:22.44 | *** part/#asterisk Carlos_PHX (n=carlos@ip68-108-193-174.ph.ph.cox.net) |
| 15:23.05 | [TK]D-Fender | Naikrovek: Well you seem to place blame first, and then proof aspires to second place. |
| 15:23.16 | Naikrovek | fine! fuck me .. i take it back |
| 15:23.21 | Naikrovek | asterisk is my lord |
| 15:23.24 | Naikrovek | i bow to asterisk |
| 15:23.27 | Naikrovek | are you FUCKING happy now |
| 15:23.31 | Naikrovek | cripes |
| 15:23.50 | [TK]D-Fender | Naikrovek: No, I'd simply like to see your issue. |
| 15:24.17 | [TK]D-Fender | Naikrovek: I'm not asking for undo praise of * or anything, simply that you not take blind pot-shots |
| 15:24.31 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 15:24.34 | [TK]D-Fender | Naikrovek: And if you have an issue many of us are simply here to help. |
| 15:30.35 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 15:30.53 | Corydon76-dig | wopsy: I have no experience with LDAP, but have you tried the schema that is in the SVN branch? |
| 15:31.12 | manxpower | ARGH! We just lost power |
| 15:31.47 | murdock_ut | manxpower: Release the gerbils. |
| 15:33.02 | leifmadsen | manxpower: where are you? |
| 15:33.53 | wopsy | hmmmm no Corydon76-dig |
| 15:33.59 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 15:34.19 | wopsy | but where i can found it ? http://svn.digium.com/svn/asterisk/branches/1.6.1/ ? |
| 15:34.39 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:35.04 | Corydon76-dig | wopsy: Yes, in contrib/scripts |
| 15:35.58 | wopsy | http://svn.digium.com/svn/asterisk/branches/1.6.1/contrib/scripts/asterisk.ldap-schema ok ok |
| 15:36.05 | wopsy | Thanks a lot |
| 15:36.15 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:36.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:36.30 | wopsy | gonna test it tomorrow |
| 15:36.32 | Corydon76-dig | wopsy: and the corresponding asterisk.ldif is in the same directory |
| 15:37.20 | Corydon76-dig | wopsy: from what I understand, you have to use a matched pair for it to work |
| 15:37.35 | manxpower | power back on |
| 15:37.40 | manxpower | hugs his battery backup |
| 15:37.47 | Qwell | manxpower: heh |
| 15:37.57 | Qwell | manxpower: our ISP was hit too |
| 15:38.01 | Qwell | what happened? O.o |
| 15:38.19 | Corydon76-dig | What, no diesel generator? |
| 15:38.32 | wopsy | a matched pair ? |
| 15:38.33 | murdock_ut | Heck with diesel, go natural gas. |
| 15:38.44 | coppice | fart power |
| 15:39.18 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 15:39.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:39.29 | manxpower | Qwell: HSV? |
| 15:39.42 | Qwell | manxpower: yeah |
| 15:39.54 | manxpower | Qwell: No idea. |
| 15:40.21 | Corydon76-dig | wopsy: yeah, the ldif and ldap-schema pair |
| 15:40.31 | manxpower | Qwell: what ISP? |
| 15:40.45 | Qwell | manxpower: umm, that one. you know |
| 15:41.04 | Corydon76-dig | The big one |
| 15:41.07 | murdock_ut | AOL??? hehehehe |
| 15:41.10 | manxpower | There's one like 3 blocks from where I live. I don't recall the name, but would recognize it. |
| 15:41.16 | *** join/#asterisk djin (n=djin@91.212.183.66) |
| 15:41.16 | Qwell | murdock_ut: yeah that's them |
| 15:41.20 | Corydon76-dig | also responsible for earthquakes in California |
| 15:41.49 | wopsy | Haa ok Corydon76-dig :) I already use ldap-schema + ldif but i think i do a mistake with the ldif from voip.org ^^ |
| 15:41.58 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 15:42.04 | manxpower | wopsy: much of the info on voip-info is WRONG. |
| 15:42.23 | Corydon76-dig | wopsy: when they're updated, they both tend to be updated at the same time |
| 15:42.31 | Qwell | manxpower: oh, right. APID |
| 15:42.33 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 15:42.56 | Corydon76-dig | wopsy: which is why I suggest that you update both at the same time, from the same source |
| 15:43.02 | wopsy | y but im a noob ! i work as trainee since 1 month and i try to install and configure asterisk with billing , ldap etc etc .. |
| 15:43.15 | manxpower | Qwell: Ah. I hate APID. They did not offer me a job. |
| 15:47.52 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:50.50 | Katty | has just sent a letter to UPS. |
| 15:51.23 | rob0 | just kicked his UPS (but not the power switch, thankfully!) |
| 15:51.46 | Katty | i would encourage you to send letters to corporations more often. |
| 15:52.15 | Katty | you'd be surprised the impact they make. |
| 15:52.19 | rob0 | Le corporacion, c'est moi! |
| 15:52.33 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-68-246.link.net.pk) |
| 15:52.36 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:52.53 | Qwell | Katty: I did that once, and they shipped me some potted plant and fruit basket thing. |
| 15:53.06 | Qwell | they even used another carrier to ship it O.o |
| 15:53.11 | Katty | hehe. |
| 15:53.20 | Katty | but it still means they read it. |
| 15:53.36 | Qwell | there was also a short apology letter with it |
| 15:54.05 | Katty | i'm going to find two other companies to send positive letters to today. |
| 15:54.12 | *** join/#asterisk FinboySlick (n=shark@207.134.8.4) |
| 15:55.16 | *** join/#asterisk dajhorn (n=dajhorn@transmisor.vanadac.com) |
| 15:56.01 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 15:56.31 | FinboySlick | I need to create a very very simple asterisk config that just 'converts' sip to iax, some sort of a bridge. Are there examples of such things? Does it require an actual dialplan or is there some specific functionality to just 'bridge' channels? |
| 15:57.34 | manxpower | FinboySlick: exten => _X.,1,Dial(IAX2/${EXTEN}@iax2peer) |
| 15:58.51 | FinboySlick | Okay, so it's pretty much just a matter of doing this and the reverse case, right? |
| 15:58.51 | leifmadsen | FinboySlick: what you're asking for is exactly what asterisk does -- it bridges channels together, regardless of technology |
| 15:59.51 | FinboySlick | leifmadsen: Well, that was my understanding too. I just wanted to make sure I was doing it right. |
| 16:00.28 | leifmadsen | sometimes you should just try :) |
| 16:01.03 | LiNeTuX_rem | <leifmadsen> Do or do not. There is no try. |
| 16:01.24 | leifmadsen | tries not to kill LiNeTuX_rem |
| 16:01.26 | LiNeTuX_rem | :) |
| 16:01.28 | leifmadsen | ;) |
| 16:02.38 | FinboySlick | Heh, I'm not afraid to try, I just think it's important to do it properly. I don't like hacking something together when there's a 'proper' way of doing things. Obviously in this case the proper way is my envisioned hack but checking with knowledgeable people never hurts. |
| 16:03.07 | leifmadsen | fair enough |
| 16:03.33 | leifmadsen | I'm just jaded because too many people come in here asking a question that could be simply answered by actually trying first |
| 16:06.26 | leifmadsen | kicks softphones that modify the audio volumes on his laptop when they start up! |
| 16:07.54 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 16:19.59 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:21.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:22.12 | *** join/#asterisk dominic1 (n=Miranda@213.221.82.242) |
| 16:22.25 | dominic1 | hello guys |
| 16:23.10 | dominic1 | short question..... I am looking for a ISDN mediagateways ( BRI AND/OR PRI ) for Asterisk and possibly for MS Office Communication server |
| 16:23.25 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 16:23.44 | Qwell | dominic1: Why not just get a PRI (or BRI) card? |
| 16:24.47 | dominic1 | Cause i am using virtualized asterisk and MS communication server systems..... and MS communication Server is using SIP over TCP |
| 16:26.16 | shido6 | yikes |
| 16:26.30 | shido6 | u can make a mediagateway |
| 16:26.34 | shido6 | with a mini itx |
| 16:26.36 | shido6 | and bri card |
| 16:28.21 | dominic1 | Yes, but i am looking for the version, which everybody can control. Like something with a small webgui to create new SIP or IAX trunks. I have some MS n00bs here :-( |
| 16:28.49 | Qwell | dominic1: Do you *really* want that type of person controlling critical infrastructure? |
| 16:28.53 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 16:29.29 | bmoraca | EVERYONE creating SIP or IAX trunks? yikes |
| 16:29.54 | SuPrSluG | dominic1: audiocodes has gateways |
| 16:29.56 | [TK]D-Fender | dominic1: AudioCodes Median |
| 16:29.57 | dominic1 | it's our testing infrastructure, for production we have other systems... |
| 16:29.58 | [TK]D-Fender | dominic1: AudioCodes Mediant |
| 16:29.59 | manxpower | this is too painful to watch |
| 16:30.03 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 16:30.06 | SuPrSluG | :-) |
| 16:30.40 | coppice | test with one thing. run another. yep, that sounds like a good idea :-) |
| 16:31.00 | bmoraca | doesn't asterisk 1.6 support SIP over TCP? |
| 16:31.17 | Qwell | bmoraca: yes |
| 16:31.19 | dominic1 | yes, but not marked as stable.... |
| 16:31.23 | dominic1 | i think.... |
| 16:31.33 | Qwell | dominic1: OCS isn't exactly stable either. |
| 16:31.42 | bmoraca | hah! |
| 16:32.16 | bmoraca | i wanted to test out OCS, but when I found that just to INSTALL it, you need to make drastic changes to your AD tree, I said fuck it, it's not worth it. |
| 16:32.24 | *** join/#asterisk jkat (n=asdf@mx.wunderkindpr.com) |
| 16:32.31 | Qwell | bmoraca: I would have stopped at "AD?" |
| 16:32.46 | jkat | hello all, need help again... |
| 16:33.03 | jkat | having issues with dahdi configuration i guess |
| 16:33.15 | [TK]D-Fender | OCS + AD = OCD? |
| 16:33.23 | jkat | maybe my dialplan... |
| 16:33.32 | jkat | i really have no clue |
| 16:34.07 | SuPrSluG | neither do we. try explaning the issue |
| 16:34.11 | jkat | lol |
| 16:34.21 | jkat | ok i get everyone is busy (1:0/0/1 |
| 16:34.50 | bmoraca | jkat: can't help what we don't know...an error would be helpful...or a failed call log...or SOMETHING |
| 16:34.54 | jkat | can you tell me what the 1:0/0/1 |
| 16:34.57 | jkat | means |
| 16:35.11 | jkat | i assume thats trunk 1, channel 1? |
| 16:35.18 | TSM | OCS only good if you have a good AD plan, just wait till one thing goes wrong with the AD, you will loose all computers/phones, the bosses would love you :) |
| 16:35.53 | *** join/#asterisk intralanman (n=lanman@va-67-76-163-226.sta.embarqhsd.net) |
| 16:36.25 | dominic1 | If something in you AD goes wrong, you will have more problems than UMS ...... |
| 16:36.40 | jkat | i think i've got chan_dahdi.conf, zapata.conf setup right |
| 16:37.02 | jkat | dahdi show status says the status is ok |
| 16:37.07 | jkat | before it was showing unconfigured |
| 16:37.24 | jkat | i fixed that by disabling all the other modules I didnt need |
| 16:37.29 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:37.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:38.08 | jkat | dahdi show channel 1 gives me a lot of stuff |
| 16:38.39 | SuPrSluG | show cli out put of a call |
| 16:38.45 | SuPrSluG | ~pastebin |
| 16:38.46 | infobot | [~pastebin] 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 |
| 16:39.02 | [TK]D-Fender | jkyou should not be needing or using zapata.conf |
| 16:39.04 | jkat | ah thank you... was wondering where to paste all that stuff...lol |
| 16:39.22 | jkat | ok zapata.conf came from the old install i just put there for reference |
| 16:40.10 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 16:41.02 | jkat | http://pastebin.com/mfe229cc |
| 16:42.08 | shido6 | Zap? what version of asterisk are you using? |
| 16:42.17 | jkat | asterisknow |
| 16:42.22 | jkat | with tdm400p |
| 16:43.19 | SuPrSluG | looks like you're trying to dial 111? and the 9 isn't strippped? |
| 16:44.28 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 16:44.49 | jkat | well its a FXO plugged into an existing telephone system |
| 16:45.04 | jkat | i was trying to dial extension 111 on the existing system |
| 16:45.34 | jkat | i thought the 9 was being stripped |
| 16:45.47 | SuPrSluG | :1 w/ dial |
| 16:45.57 | Qwell | it was stripped.. |
| 16:46.55 | jkat | its like it isnt getting a dial tone |
| 16:47.19 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 16:47.24 | Qwell | you're dialing to another pbx...enblock. why would you expect a dialtone? |
| 16:47.58 | jkat | true, shouldnt get dial tone unless i dial 99 and an outside number |
| 16:47.59 | ariel_ | afternoon folks |
| 16:48.27 | jkat | but i am getting everyone is busy/congested... |
| 16:48.32 | *** join/#asterisk SRecall (n=info@64.62.233.194) |
| 16:48.44 | Naikrovek | jkat: that's the unreachable sound, not "busy" |
| 16:48.49 | jkat | ok |
| 16:49.03 | jkat | so i think my dahdi configs are off |
| 16:49.11 | Naikrovek | so think of them as being unreachable and you'll get a bit of a clue as to what's going on in most cases |
| 16:49.36 | jkat | ok |
| 16:50.16 | jkat | when i run dahdi show channel 1 i get "hookstate: offhook" at the bottom |
| 16:51.04 | jkat | lsdahdi shows 1 FXS FXSKS (in use) |
| 16:51.49 | jkat | but port=1,FXO under dahdi_scan |
| 16:52.12 | jkat | so i think I've got a FXO configured as FXS? or am I way off base |
| 16:52.49 | jkat | well iknow i've got an fxo... |
| 16:52.56 | *** join/#asterisk btsteve (n=tstevens@204.10.20.30) |
| 16:54.41 | jkat | should /etc/dahdi/system.conf have fxols=1 ? |
| 16:55.16 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 16:55.25 | jkat | ... nvm |
| 16:55.37 | jkat | fxo uses fxs signaling... |
| 16:58.11 | jkat | my chan_dahdi.conf ... http://pastebin.com/m33b152a0 |
| 16:58.32 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 16:59.01 | jkat | http://pastebin.com/m6adfbc61 dahdi/system.conf |
| 16:59.03 | [TK]D-Fender | jkWhat color of module is on your card? |
| 16:59.06 | jkat | red |
| 16:59.09 | [TK]D-Fender | jkat: What color of module is on your card? |
| 16:59.50 | [TK]D-Fender | jkaOk, configs look fine for that.jkat: |
| 17:00.07 | jkat | so maybe my dialing rules? |
| 17:00.25 | jkat | what other configs can i show you? |
| 17:00.36 | [TK]D-Fender | jkat: Youshould be doing Dial(DHADH/1/thenumberhere) |
| 17:00.38 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 17:01.03 | Qwell | [TK]D-Fender: way to butcher that |
| 17:01.06 | [TK]D-Fender | jkat: Do "dahdi show status", "dahdi show channels", "core show channels" |
| 17:01.13 | [TK]D-Fender | Qwell: YESH |
| 17:01.38 | [TK]D-Fender | jkcorrecting my obvious massacring of DAHDI there... |
| 17:01.45 | [TK]D-Fender | STUPID AUTOCOMPLETE |
| 17:01.48 | [TK]D-Fender | gah |
| 17:02.01 | jkat | http://pastebin.com/m6454cd0a |
| 17:03.08 | [TK]D-Fender | jkat: You will also want to verify the PORT the cord is plugged into. they may not be ordered the way you think they are. |
| 17:03.21 | jkat | plugged into the one with the green light on |
| 17:03.58 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:04.07 | jkat | Dial("IAX2/iaxy-3468", "Zap/1/1111") are you saying this should be Dial("IAX2/iaxy-3468", "DAHDI/1/1111")? |
| 17:04.37 | [TK]D-Fender | jkat: PB a complete call attempt and do "core set debug 10" first |
| 17:05.41 | *** part/#asterisk larin (n=larin@mail.cs-service.by) |
| 17:05.47 | jkat | http://pastebin.com/m60015db2 |
| 17:07.10 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 17:07.33 | [TK]D-Fender | jkat: Put an analog phone in parallel witha splitter and listen in on the attempt |
| 17:07.44 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:08.16 | jkat | ah ha... i see that in the dialplan |
| 17:08.30 | jkat | 's' => 1. Set(${ARG1}=${EXTEN}) [pbx_config] |
| 17:08.30 | jkat | <PROTECTED> |
| 17:08.40 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 17:08.58 | jkat | yeppers |
| 17:09.03 | SuPrSluG | ~dahditools |
| 17:09.04 | tzafrir_laptop | anybody knows what is "C7" as a way to pass a caller ID string? |
| 17:09.04 | muiro | with MeetMe(), is there any way I can check to see if a conference has any users in it? |
| 17:09.21 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:09.31 | tzafrir_laptop | Maybe something that is used in India |
| 17:09.32 | jkat | wwwhere is the dialplan config file? |
| 17:09.54 | tzafrir_laptop | extensions.conf (or .ael , but that's something slightly different) |
| 17:10.10 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
| 17:10.11 | dmz | hey y'all, i have a script that drops .call files into /var/spool/asterisk/outgoing and occasionally (at least weekly) it doesn't pickup the files that are put in there (using mv, not cp) but if i restart asterisk it processes the calls. any suggestions? |
| 17:10.17 | SuPrSluG | jkat: dahdi tools |
| 17:10.42 | *** join/#asterisk errotan (n=errotan@5403E426.catv.pool.telekom.hu) |
| 17:11.15 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:11.19 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:15e1:43be:6078:eeda) |
| 17:11.22 | shido6 | weird |
| 17:11.25 | Skeeter- | what is the most stable platform for asterisk |
| 17:11.38 | shido6 | what user do u have running asterisk, dmz? |
| 17:11.55 | dmz | asterisk, and the call file chown's the file to asterisk user before moving it in |
| 17:12.18 | shido6 | what user owns the /var/spool/asterisk and folders underneath |
| 17:12.27 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:12.29 | shido6 | that came out wrong - but u get the idea |
| 17:12.45 | dmz | asterisk |
| 17:12.46 | tzafrir_laptop | Skeeter-, the floor |
| 17:12.52 | shido6 | lol |
| 17:12.52 | dmz | it's a default debian build |
| 17:13.12 | shido6 | what user owns the /var/spool/asterisk folder and the folders beaneath it |
| 17:13.18 | shido6 | bean? |
| 17:13.23 | shido6 | i think i have taco bell on the brain |
| 17:13.29 | shido6 | was headed out to get some |
| 17:13.43 | shido6 | so now Im just cracking up like a hard taco shell... |
| 17:13.55 | dmz | :) |
| 17:14.10 | shido6 | dmz check the permissions |
| 17:14.28 | shido6 | what happens when u cp ? |
| 17:14.35 | jkat | should DAHDI always be caps? |
| 17:14.37 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:15.02 | jkat | as in trunk = DAHDI/g1 |
| 17:16.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:16.50 | jkat | thank god for #asterisk! |
| 17:17.06 | jkat | stupid name change from zapata to dahdi |
| 17:17.11 | jkat | :-P |
| 17:17.33 | beek | jkat: DAHDI is an acronym, so it's capitalized. |
| 17:17.37 | *** join/#asterisk ingenius (n=alektro@201.254.127.196) |
| 17:17.43 | beek | jkat: The name change was required. |
| 17:18.53 | rob0 | Thank the lawyers, like so many other stupid things happening in this world. |
| 17:19.10 | jkat | oh yeah i read all about it |
| 17:19.17 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:19.24 | jkat | let me rephrase to say... stupid trademarks |
| 17:19.27 | jkat | etc |
| 17:19.41 | [TK]D-Fender | jkat: case doesn't amtter for the tech in Dial() |
| 17:20.15 | jkat | thank you again |
| 17:20.20 | jkat | i'm dialing and receiving on iax now |
| 17:23.05 | SRecall | Anyone know if the followme app has realtime support? |
| 17:23.05 | dmz | shido6 you don't 'cp' into outgoing, i don't have exact reference but for automated calls mv is best (instant reference change vs having to copy files & access times /etc) |
| 17:23.17 | dmz | i'll try changing delay between moving of files in |
| 17:23.27 | leifmadsen | SRecall: I don't think so -- I just use the dialplan and func_odbc |
| 17:23.42 | Katty | oh. |
| 17:23.47 | Katty | and here i thought i had messages waiting for me. |
| 17:23.56 | Katty | pouts. |
| 17:24.18 | [TK]D-Fender | dmz: Don't |
| 17:24.48 | [TK]D-Fender | dmz: if * isn't taking them, either its a permissions issue, timestamp, or you don't have pbx_spool loaded |
| 17:25.21 | [TK]D-Fender | dmz: they must be "mv"'d in, not cp'd because * will try to instantly lock it and havoc will ensue |
| 17:28.03 | dmz | D-Fender, no it's taking the calls, but occasionally it's not making a few of them and we're just getting .call files left in outgoing |
| 17:28.11 | dmz | i have to restart * to get it to process them |
| 17:29.12 | jkat | no sip calls arent going out... http://pastebin.com/m72a99e8d |
| 17:29.12 | dmz | hmm this may have something to do w/it : asterisk[1101]: segfault at 00000000000000f8 rip 00002aaaab2edbfe rsp 00000000401dc790 error 6 |
| 17:29.19 | jkat | *now sip calls... |
| 17:30.09 | leifmadsen | dmz: honestly -- use the Originate action in the asterisk manager if you need to rely on triggering calls -- .call files are not very reliable, especially for any sort of volume |
| 17:30.48 | dmz | leifmadsen i'll have to check on that, i've not done anything w/asterisk manager; i assume it has api interface so i can just integrate my perl scripts? |
| 17:31.08 | leifmadsen | dmz: there are several examples, and you're not going to be monitoring anything, so it should be pretty straight forward |
| 17:31.25 | leifmadsen | dmz: its just stdin/stdout via a socket I'm pretty sure |
| 17:31.47 | leifmadsen | so ya -- you can just modify your perl scripts to use it instead of generating a file |
| 17:31.57 | dmz | i'll check it out, thanks |
| 17:32.12 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 17:32.12 | leifmadsen | np! http://astbook.asteriskdocs.org should have a perl example if I remember correctly... |
| 17:32.21 | dmz | we're tying it into our zabbix system to get outbound alerts, so people get upset when they don't get an alert notification :) |
| 17:32.39 | leifmadsen | dmz: ya -- the manager is definitely more appropriate to what you want it sounds like |
| 17:32.44 | leifmadsen | it'll be a lot more reliable |
| 17:33.15 | *** join/#asterisk davidandgoliath (n=David@S0106001d60d4e488.vn.shawcable.net) |
| 17:36.28 | dmz | cool, i'll check it out. thanks |
| 17:36.51 | leifmadsen | np! |
| 17:37.30 | raden_work | how well does VOIP work over WIFI ? |
| 17:37.37 | raden_work | like as a main internet connection |
| 17:37.46 | raden_work | 1mb / 1mb 2 mile link |
| 17:37.52 | dmz | another question :) lets say i have multiple data-centers that I want to run central * boxes on for multiple offices all trunked together; is there any opensource solution (or one i can build) to fully integrate failover / redundancy at a core switch level |
| 17:38.10 | dmz | raden_work 1M should handle voip fine, depending on the # of calls @ once |
| 17:38.19 | dmz | it's not a distance but a bandwidth issue |
| 17:38.21 | raden_work | 3 -5 calls max |
| 17:38.29 | raden_work | my question is stability and packet jitter |
| 17:40.14 | leifmadsen | raden_work: depends on the wifi connection |
| 17:40.20 | leifmadsen | raden_work: works great in my apartment |
| 17:40.53 | raden_work | ikm talking a wifi bridge with tranzeo radios to a T3 |
| 17:41.35 | bmoraca | raden_work: every environment is different, and it depends on the particular wifi connection being used |
| 17:41.50 | bmoraca | it could work perfectly, and it could completely bomb |
| 17:42.21 | leifmadsen | it really has nothing to do with VoIP and everything to do with the quality of the connection |
| 17:42.50 | raden_work | well my only concern is we would have a clear line of sign to the radio a top the water tower then that radio links to another water tower 8 miles away and then to the T3 |
| 17:43.00 | raden_work | so its making 2 wifi hops |
| 17:43.32 | bmoraca | raden_work: again, it simply depends. if it's a good link with low packet loss, low retransmissions, and low latency, you should be OK. |
| 17:44.07 | *** join/#asterisk clyons (n=clyons@unaffiliated/clyons) |
| 17:44.21 | raden_work | well out ADSL sucks :( |
| 17:44.34 | raden_work | we loose registration a few times a hour |
| 17:44.46 | bmoraca | that's probably not your ADSL. |
| 17:44.54 | edguy3 | raden_work: take a look at Brian Capouch's network.. multiple wifi hops all around the corn fields of indiana. |
| 17:44.55 | raden_work | what you think it is ? |
| 17:44.56 | bmoraca | that's probably your NAT router. what model do you have? |
| 17:45.11 | raden_work | FVX538 |
| 17:45.21 | raden_work | does it with WRT54GL |
| 17:45.27 | raden_work | my Buffalo airstream |
| 17:45.29 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:45.49 | raden_work | I took asterisk box to my house threw on cable never lost registration over 24 hour period |
| 17:45.51 | bmoraca | those are all SOHO routers. trying to run a business off a SOHO router is stupid to begin with. buy a real router. |
| 17:46.13 | raden_work | were trying to get 2/4 cable service out here but they have to run a line almost a mile i dont even want to know what thats going to cost |
| 17:46.30 | raden_work | bmoraca, recomend one please |
| 17:46.33 | bmoraca | switching ISPs won't help if the issue is your router |
| 17:46.40 | bmoraca | raden_work: Cisco ASA5505 |
| 17:47.02 | raden_work | would i be better running my linux box as the router ? |
| 17:47.07 | bmoraca | no |
| 17:47.28 | raden_work | wow same price we paid for netgear |
| 17:47.54 | bmoraca | i have hosted PBX customers running on SOHO routers, and it pains me every time. i have a 7 phone one on a netgear rangemax...thankfully no rpoblems, but still |
| 17:48.08 | bmoraca | raden_work: how many users do you have, and out many external registrations? |
| 17:48.18 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:48.38 | raden_work | 3 users |
| 17:48.46 | raden_work | 1 external registration through router |
| 17:48.55 | raden_work | will be 2 once i can get teliax setup |
| 17:49.22 | bmoraca | interesting |
| 17:49.35 | raden_work | whats interesting > |
| 17:50.17 | bmoraca | shouldn't have issues with so few...but, anything's possible |
| 17:50.36 | raden_work | loose registration with callcentric just having everything idle |
| 17:50.41 | raden_work | drives me nutz |
| 17:51.02 | *** join/#asterisk wopsy (n=80475@AToulouse-754-1-1-6.w90-11.abo.wanadoo.fr) |
| 17:52.55 | raden_work | how would we handle dual wans with cisco |
| 17:53.27 | dmz | what's to handle, outbound access or inbound availability? |
| 17:54.27 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 17:55.19 | raden_work | we will have 2 Internet connections how can we have both available to network without having dual wans ? |
| 17:56.41 | bmoraca | why do you have two internet connections, and how are they currently being used? |
| 17:57.17 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 17:57.33 | dmz | you can use something like balance-ng (google it) to load balance outgoing traffic & ensure outbound access is always available (assuming you don't have both through same local loop and a local telco problem takes out both lines) |
| 17:58.58 | raden_work | one is wifi which they still dont have up and the other one is DSL |
| 17:59.06 | raden_work | going to completly diffrent places |
| 17:59.18 | Naikrovek | is that to increase your bandwidth? why did you get two |
| 17:59.33 | dmz | as long as it's outbound access your trying to balance it's just a routing issue (could even use ibgp) |
| 17:59.47 | Naikrovek | dmz: if the ISPs will cooperate |
| 17:59.49 | bmoraca | raden_work: what is the purpose? is one a connection to a separate private network? if so, that's fairly easily done with static routes. |
| 18:00.04 | raden_work | both are WAN to internet |
| 18:00.13 | bmoraca | raden_work: for what purpose? |
| 18:00.15 | Naikrovek | the ASA5505 (w/Security Plus license) can do failover automatically if that's what you want |
| 18:00.27 | raden_work | incase DSL goes down wifi for backup |
| 18:00.29 | Naikrovek | did you get dual connections to increase overall bandwidth? |
| 18:00.34 | Naikrovek | okay failover |
| 18:00.36 | dmz | Naikrovek, ibgp is internal telling your internal routers where to route through |
| 18:00.47 | Naikrovek | the ASA5505 (w/Security Plus license) can do failover automatically if that's what you want |
| 18:00.49 | dmz | Naikrovek and you can send your outgoing traffic anywhere you want; just NAT it and send it out |
| 18:00.52 | Naikrovek | dmz: ah i missed the 'i' |
| 18:00.52 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 18:00.55 | dmz | :) |
| 18:01.06 | dmz | now if he wants inbound access that's another story |
| 18:01.07 | raden_work | Naikrovek, i was originally going to run all VOIP traffic on DSL and everything else over WIFI but yeah |
| 18:01.29 | citywok | on a call redirect, i am unable to pass variables with the command. they dont attach to the new channel. is there a known way to do this? upon googlign i've found nothing. |
| 18:01.33 | raden_work | Cisco ASA 5505 is a switch |
| 18:01.38 | raden_work | how can i use that as a router ? |
| 18:01.44 | Naikrovek | it's a lot more than a switch |
| 18:01.52 | bmoraca | raden_work: inbound traffic will not work under this scenario...however, the security firmware can do failover |
| 18:02.11 | Naikrovek | the security license is spendy tho |
| 18:02.13 | bmoraca | raden_work: it's a stateful firewall (what you know as a router) with an integrated semi-managed switch |
| 18:02.15 | Naikrovek | i just bought it for our 5505 |
| 18:02.34 | bmoraca | Naikrovek: cheaper to buy the bundle from the beginning than to pay for the upgrade |
| 18:02.38 | Naikrovek | it's a bad-ass firewall, it's a NAT/PAT/DHCP device |
| 18:02.55 | Naikrovek | bmoraca: yes, but i got the upgrade anyway since i had no hand in the original purchase |
| 18:03.04 | bmoraca | ahh |
| 18:03.14 | dmz | bmoraca, stateful fw != router :) |
| 18:03.17 | raden_work | how would i hookup 2 wans to it ? |
| 18:03.31 | Naikrovek | you're confusing the term "wan" with "internet connection" |
| 18:03.39 | Naikrovek | and it's easy to set up with the security plus license |
| 18:03.45 | bmoraca | dmz: i know that, and you know that, but most people see "router" on the linksys box and don't realize that it's not really a router |
| 18:03.47 | Naikrovek | there's a wizard for it on the device if i recall |
| 18:03.48 | dmz | personally i use linux w/fwbuilder doing my own heartbeat & session handling and it costs me nothign; then i virtualize it & i have up to 800 firewalls in 1U :) |
| 18:04.45 | dmz | actually my goal is 1024 firewalls in 1u (we have lots of customers who use firewalls) :D |
| 18:04.57 | bmoraca | raden_work: a "wan" connection is simply a connection to a device which bridges you on to a larger network. the switch ports on the ASA can be configured to do a lot of things. and, as he said, i believe there's a wizard for doing failover |
| 18:04.59 | raden_work | ok can i have 2 internet connections on the cisco without spending a fortune ? |
| 18:05.12 | Naikrovek | raden_work: the ASA is like a super switch/firewall/VPN endpoint. it's got a Cisco IOS and you can put a different subnet on each port if you want, and each can have their own vlan as well |
| 18:05.19 | *** join/#asterisk EiN` (n=einstein@unaffiliated/einstein/x-615171) |
| 18:05.29 | raden_work | and do theyt have one with like 16 ports ? |
| 18:05.32 | bmoraca | raden_work: no, but if you want things to work right, you shouldn't be concerned about cost |
| 18:05.33 | dmz | you can configure them with failover and a single connection each, that way only 1 is used but if one fails it fails over to use the 'backup' |
| 18:05.41 | Naikrovek | raden_work: the ASA 5505 (10-user) w/security plus license is something like $600 if i recall |
| 18:05.45 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 18:05.52 | dmz | it's not a matter of ports but a matter of hot-hot vs hot-warm and what type of connections you can get w/what your trying |
| 18:05.55 | bmoraca | dmz: that's device failover, yes, but there's also WAN failover as well |
| 18:06.00 | raden_work | Cisco ASA 5505 10-user Bundle $347.00 |
| 18:06.14 | Naikrovek | that doesn't include the security plus license |
| 18:06.15 | dmz | well wan failover is still only hot-warm |
| 18:06.16 | Naikrovek | i don't think |
| 18:06.19 | raden_work | Cisco ASA 5505 Firewall Edition Bundle - security appliance $739.00 |
| 18:06.26 | Naikrovek | that's it |
| 18:06.37 | raden_work | whats the security plus do ? |
| 18:06.48 | dmz | if you have 2 asa's & a load balancer (balance ng rocks and isn't too $$) then you can load your traffic across each connection depending on the capacity and actually increase throughput |
| 18:06.49 | raden_work | boss will never go for that :( |
| 18:07.08 | Naikrovek | internet connection failover, provides up to 20 individual local networks as opposed to two (so no DMZ without security plus) |
| 18:07.15 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-70-60.link.net.pk) |
| 18:07.41 | dmz | 2 local networks is "office & dmz" vs 20 different networks and i think it just adds vlan support, not 20 interfaces |
| 18:07.50 | bmoraca | you can do a DMZ without the security bundle...it just doesn't work very well |
| 18:08.01 | manxpower | The first thing you should do with a Cisco firewall is disable the SIP support. |
| 18:08.23 | dmz | the 2nd thing is swap it out for a bsd or linux box ;) |
| 18:08.23 | Naikrovek | without security plus you get two networks "inside" and "outside" if you want DMZ to be fully accessible by both inside and outside you need the security plus license |
| 18:08.34 | bmoraca | manxpower: depends on who you talk to...Adtran's devices don't have the ability to NAT a SIP connection and rely on an ALG :P |
| 18:08.52 | Naikrovek | raden_work: we use an asa5505 and it's working out quite well for us. |
| 18:09.26 | dmz | you can always try a vyatta solution |
| 18:10.40 | bmoraca | honestly, having multiple WANs is pretty pointless. chances are that if one of them is down, there's a good chance that the other will be down as well |
| 18:11.20 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 18:11.22 | dmz | and if dsl is down typically hardline phones may be having problems too; so just take the outage as a time to take a lunch break |
| 18:12.05 | *** join/#asterisk jackal (n=jackal@pool-71-171-99-11.clppva.fios.verizon.net) |
| 18:12.18 | bmoraca | if you really need multiple WANs, you should register an AS number and engage in BGP with your two ISPs. that's the best way to do it :P |
| 18:12.38 | Naikrovek | bmoraca: true dat |
| 18:13.08 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:16.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:18.08 | raden_work | dmz, i have linux box that asterisk is on easily make it into a router |
| 18:19.02 | raden_work | bmoraca, the Internet connections go 2 complete opposite ways |
| 18:19.19 | bmoraca | raden_work: as far as the internet is concerned, there is no such thing as "opposite ways" |
| 18:19.40 | raden_work | I understand that |
| 18:19.42 | bmoraca | raden_work: additionally, it's not a good idea to put asterisk on a box that has two paths outside of the local network. |
| 18:20.06 | raden_work | why is that ? |
| 18:20.13 | bmoraca | RTP path issues |
| 18:20.24 | raden_work | that i can understand |
| 18:20.38 | raden_work | If i could get cable out here id be set |
| 18:21.17 | bmoraca | raden_work: like i said, it's unlikely that your DSL is the issue, unless your modem is incorrectly configured or failing, and if it were failing, you'd be noticing other issues |
| 18:21.32 | bmoraca | i have shit-tons of customers on DSL and they all work just fine |
| 18:21.46 | manxpower | The real problem is you are talking about sending mission critical phone calls over the INTERNET. |
| 18:21.55 | raden_work | our pack jitter is through the roof as well |
| 18:21.58 | manxpower | If you want reliable get a direct link to an ITSP or use PSTN lines. |
| 18:22.16 | raden_work | PSTN = $60 a line x 5 lines |
| 18:22.19 | raden_work | do the math |
| 18:22.26 | raden_work | they rape us here |
| 18:22.42 | manxpower | raden_work: and how much will it cost for a day or two of downtime? |
| 18:23.11 | manxpower | raden_work: Where is "here"? |
| 18:23.17 | raden_work | wisocnsin |
| 18:23.27 | raden_work | centurytel sucks for business services |
| 18:23.36 | *** join/#asterisk Shaun2222 (n=Shaun222@ip68-5-154-233.oc.oc.cox.net) |
| 18:23.37 | raden_work | DSL + 1 Line costs us $120 |
| 18:23.46 | raden_work | plus 3.5 cents a min |
| 18:23.48 | manxpower | Just understand, no matter what you do it will not be as reliable as the PSTN |
| 18:24.06 | raden_work | I understand that but on cable at home ive never had it drop |
| 18:24.10 | manxpower | I recommend a couple of local lines + SIP to an ITSP CenturyTel peers with. |
| 18:24.19 | *** join/#asterisk chocoz_ (n=alex@pool-173-56-225-163.nycmny.east.verizon.net) |
| 18:24.21 | jkat | thanks again guys |
| 18:24.29 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 18:24.48 | raden_work | trying to eliminate centurytel |
| 18:26.10 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:27.45 | bmoraca | we're getting a wimax connection at an upcomming tradeshow...hoping that we can make voip calls out of it for demo purposes...will be fun |
| 18:28.04 | chocoz_ | hi all |
| 18:28.47 | chocoz_ | im trying to run some benchmarks on codec transcoding, but i cant seem to force asterisk to use the codecs i want. |
| 18:28.56 | chocoz_ | does anyone have any experience with this? |
| 18:30.31 | Shaun2222 | going from 1.6.0.x to 1.6.1 any big changes that may mess with me |
| 18:31.11 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 18:31.13 | spck | ls |
| 18:31.16 | spck | oops |
| 18:31.26 | spck | rm -rf / |
| 18:32.20 | manxpower | ~answers |
| 18:32.21 | infobot | answers is, like, Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 18:34.03 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:37.29 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 18:37.33 | Naikrovek | yay infobot is back |
| 18:37.41 | Naikrovek | didn't notice his/her/it's return |
| 18:37.58 | Naikrovek | ooh i gotta watch my apostrophe usage |
| 18:38.07 | Naikrovek | slaps Naikrovek. |
| 18:38.37 | raden_work | watchguard any good ? |
| 18:38.41 | raden_work | just talked to CDW |
| 18:39.06 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 18:39.37 | raden_work | anyway have to go back in 5 hours |
| 18:42.03 | shriven | Hello. I am having a problem with a call parking. The parking function seems to work fine, however I use a macro for initiating the call. In the macro I have as gotoif set to send the call to voicemail if the destination is unavailable. What is happening is that after the caller dials out with the macro, the callee answers, then the caller parks the call, the parked call slot is echoed to the caller, but then instead of the line going dead, or waitin |
| 18:42.03 | shriven | hangup, it sends the caller straight to the callee's voicemail. Any ideas what to do to avoid that? I hope that makes sense.... |
| 18:42.13 | shriven | I'm using 1.6.2 beta4 |
| 18:49.24 | *** join/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
| 18:50.02 | JerJer | anyone specifically know if by calling Hangup in the dialplan will always cause the 'h' exten to be fired? |
| 18:50.08 | JerJer | in 1.4 |
| 18:51.16 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 18:52.19 | Shaun2222 | undefined symbol: ast_smdi_interface_unref |
| 18:52.23 | Shaun2222 | anybody know what thats from |
| 18:52.46 | russellb | Shaun2222: what version |
| 18:52.53 | russellb | and do you have res_smdi.so loaded? |
| 18:53.01 | Shaun2222 | 1.6.1.x |
| 18:53.09 | russellb | there is no version called 1.6.1.x :-p |
| 18:53.13 | Shaun2222 | probably not. |
| 18:53.21 | russellb | okay, load it, that should fix it |
| 18:53.30 | Shaun2222 | i autoload=no then load only what i need |
| 18:54.44 | JerJer | Shaun2222: there are assumptions made that specific modules will exist |
| 18:55.03 | Shaun2222 | ya, i think loading res_smdi.so fixed it |
| 18:55.20 | JerJer | like res_smdi, chan_local, res_agi |
| 18:55.35 | Shaun2222 | dahdi :) |
| 18:55.47 | JerJer | who's your dahdi isn't required |
| 18:56.09 | Shaun2222 | was for me |
| 18:56.32 | JerJer | only if you require analog/digital interfaces |
| 18:57.16 | Shaun2222 | [Aug 25 11:49:42] WARNING[29675]: loader.c:417 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref |
| 18:57.16 | Shaun2222 | thats what i saw. |
| 18:57.45 | JerJer | if you don't have a need for Digium's hardware interface cards, don't load chan_dahdi |
| 18:57.56 | JerJer | i don't even build it on my systems |
| 18:58.07 | Shaun2222 | i have a t1 card from sangoma |
| 18:58.14 | JerJer | then you need it |
| 18:58.25 | JerJer | and then y ou also need smdi |
| 18:58.31 | JerJer | :) |
| 18:58.38 | chocoz_ | how can i get one asterisk box to dial another, and for the 2nd box to pick up? |
| 18:58.45 | shido6 | ...... |
| 18:58.50 | SuPrSluG | pray |
| 18:58.58 | JerJer | exten => s,1,Dial(SIP/bob) |
| 18:59.53 | SuPrSluG | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 19:00.07 | [TK]D-Fender | chocoz_: Go lookup "asterisk dual servers" on the WIKI for some inspiration. this i no different than setting up an ITSP |
| 19:00.15 | [TK]D-Fender | SuPrSluG: Good call.. |
| 19:00.29 | SuPrSluG | :-D |
| 19:00.32 | chocoz_ | ok cool, ill check into that |
| 19:00.35 | chocoz_ | thx |
| 19:01.14 | Shaun2222 | JerJer: in 1.6.0.x it wasnt.. :) |
| 19:01.25 | Shaun2222 | JerJer: also app_queue needs ast_monitor_start now i guess too |
| 19:01.45 | JerJer | i have those problems in 1.4 |
| 19:02.05 | JerJer | so its most certainly pre-1.6.0 |
| 19:03.45 | Shaun2222 | okey dokie.. no more errors in the logs, lets see if it works now... |
| 19:05.42 | Shaun2222 | bahh.. course why would it work |
| 19:09.05 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 19:09.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:09.21 | manxpower | ~siptrunk |
| 19:09.22 | infobot | extra, extra, read all about it, siptrunk is To set a SIP peer/friend/user as a trunk add either trunk=yes or wombat=yes (they both do the same thing) in the peer/friend/user definition in sip.conf |
| 19:13.04 | Shaun2222 | hmm, any known issues in 1.6.1.4 where playback in a gosub doesnt work |
| 19:13.08 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 19:15.40 | *** join/#asterisk SRecall (n=info@64.62.233.194) |
| 19:16.31 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:16.39 | SRecall | Anyone know why realtime stops working after an upgrade to 1.6.2 ? I was running 1.6.0 before. |
| 19:17.24 | asterwiki | Shaun2222: What error did you get? |
| 19:18.20 | SRecall | WARNING[18552]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk |
| 19:18.39 | SRecall | realtime status shows its connected though |
| 19:19.35 | Shaun2222 | not getting a error. |
| 19:19.46 | Shaun2222 | i have a gosub that does playback |
| 19:20.03 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:20.03 | Shaun2222 | but now with 1.6.1.4 that playback cant be heard by the user |
| 19:20.16 | Shaun2222 | the background() before that gosub they can hear though |
| 19:21.32 | leifmadsen | Shaun2222: GoSub() is really just a Goto(), so there would be no difference in the GoSub() versus just doing Goto() |
| 19:22.24 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 19:22.50 | manxpower | Shaun2222: you're not using the GoSub/Goto/Macro from inside a Dial, are you? |
| 19:23.26 | asterwiki | Shaun2222: Also check to see if maybe the file is in located somewhere else... /var/lib/asterisk/sounds or /var/lib/asterisk/sounds/en |
| 19:23.32 | manxpower | SRecall: did you read the UPGRADE*.TXT files? 1.6.0 to 1.6.2 is considered a major upgrade. |
| 19:24.28 | manxpower | 1 = major-major version, 6 = major-minor version 0/1/2 = minor-minor version |
| 19:24.41 | [TK]D-Fender | Shaun2222: Where si the comparative pastebin? |
| 19:24.48 | Shaun2222 | manxpower: yes, i am |
| 19:25.00 | *** join/#asterisk sam555 (n=chatzill@udp153808uds.hawaiiantel.net) |
| 19:25.09 | sam555 | hello all! |
| 19:25.17 | sam555 | what linux distro does asterisk use? |
| 19:25.19 | asterwiki | Shaun2222: you may need to write have something like Playback(/var/lib/asterisk/sounds/en/filename) to get it going |
| 19:25.20 | Shaun2222 | manxpower: gosub from dial |
| 19:25.28 | sam555 | i'm trying to make my asterisk install a virtual server |
| 19:25.34 | manxpower | Shaun2222: paste just the Dial line |
| 19:26.15 | manxpower | you can only collect digits using Read from inside a dial |
| 19:26.23 | Shaun2222 | oh wait |
| 19:26.26 | manxpower | I don't think even waitexten works for that |
| 19:26.34 | Shaun2222 | here, one sec |
| 19:27.08 | *** join/#asterisk phuff (n=user@sourceforge/staff/luapffuh) |
| 19:27.36 | [TK]D-Fender | [15:25]<sam555>what linux distro does asterisk use? <- whatever you install it on |
| 19:27.41 | Shaun2222 | which snip |
| 19:27.42 | Shaun2222 | http://pastebin.ca/1542174 |
| 19:27.45 | Shaun2222 | quick* |
| 19:28.37 | sam555 | [TK]D-Fender: ok, so it has to be on a distro to work? |
| 19:28.41 | sam555 | [TK]D-Fender: meaning it can't stand alone? |
| 19:28.46 | manxpower | Shaun2222: ignore everything I said. you are not using Dial |
| 19:28.50 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:28.50 | leifmadsen | sam555: see AsteriskNOW |
| 19:28.59 | sam555 | leifmadsen: k |
| 19:29.01 | Shaun2222 | manxpower: ya i know, i realized that :/ |
| 19:29.08 | leifmadsen | sam555: Asterisk is an application -- it always has to run on an OS |
| 19:29.15 | *** join/#asterisk wajasu (n=chatzill@cpe-76-187-72-59.tx.res.rr.com) |
| 19:29.15 | [TK]D-Fender | sam555: * is a piece of software, not an OPERATING SYSTEM |
| 19:29.26 | phuff | sam555: I installed Asterisk from source on my ubuntu box without a particular distro |
| 19:29.33 | leifmadsen | it's like you're asking if I can just install Word onto a freshly formatted hard drive |
| 19:29.42 | [TK]D-Fender | leifmadsen: YOU CAN"T!??! |
| 19:29.44 | sam555 | ok, gotcha |
| 19:29.56 | leifmadsen | [TK]D-Fender: you can't even buy Word in Canada anymore! :) |
| 19:29.59 | phuff | sam555: If you're not looking for particular drivers for a card it seems to work fine in any linux environment |
| 19:30.11 | [TK]D-Fender | leifmadsen: VICTORY!!!!! |
| 19:30.14 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 19:30.26 | sam555 | phuff: ok, i guess I'll install a virtual ubuntu machine and then put asterisk on that |
| 19:30.28 | leifmadsen | phuff: uhhh... you don't make sense really there. Ubuntu is a distro... |
| 19:30.34 | leifmadsen | sam555: enjoy! |
| 19:30.42 | sam555 | leifmadsen: thanks! |
| 19:30.51 | phuff | leifmadsen: I didn't install it from ubuntu packages |
| 19:31.03 | leifmadsen | phuff: yes, he was asking a totally different question though |
| 19:31.11 | phuff | leifmadsen: And it installed fine, which should be sufficient to satisfy sam555's question... |
| 19:31.29 | phuff | I thought he was asking if it needed a particular distro |
| 19:31.42 | leifmadsen | phuff: he was asking if it had to be on a distro. period. |
| 19:31.45 | leifmadsen | which of course it does. |
| 19:31.58 | phuff | Or at least a running linux machine |
| 19:32.06 | phuff | I suppose you can have a running linux machine without a distro... |
| 19:32.11 | [TK]D-Fender | not necessarily linux |
| 19:32.12 | leifmadsen | I guess.... LFS |
| 19:32.30 | leifmadsen | everyone runs some sort of distro anymore unless they are crazy masochists |
| 19:32.33 | phuff | Yeah, I didn't know if it's supported on freebds/macosx |
| 19:32.35 | asterwiki | Have you checked to see where the Playback is looking for the sound file?... maybe /var/lib/asterisk/sounds/en/... you may need to give playback the full path |
| 19:32.49 | Shaun2222 | well just downgraded to 1.6.0.13 and it works perfect... guess it's a bug. |
| 19:33.03 | [TK]D-Fender | Shaun2222: You didn't show us the failed call |
| 19:33.25 | Shaun2222 | nope. |
| 19:33.35 | Shaun2222 | dont have time for this.. revert and hopfully sombody else has time. |
| 19:35.33 | [TK]D-Fender | Shaun2222: Well ask us when you actually have the time, ability, and inclination to do something about it next time |
| 19:36.23 | Shaun2222 | [TK]D-Fender: the asterisk log wasnt showing any errors... it displayed as if it was paying a sound to the person... but it wasnt |
| 19:36.24 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:36.29 | Shaun2222 | i could hear before the gosub |
| 19:36.34 | Shaun2222 | once in the gosub no sound. |
| 19:36.41 | Shaun2222 | reverted to 1.6.0.13 and it all works fine again. |
| 19:37.35 | *** join/#asterisk eharris (i=eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 19:37.47 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 19:37.55 | [TK]D-Fender | Shaun2222: Sorry that I can't take that at face falue (for what little I can only imagine right now). |
| 19:38.04 | *** part/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 19:40.05 | asterwiki | Shaun2222: the problem may simply be the directory the file is stored in, did you check that?... this is a simple issue that shouldn't stand in the way of your upgrade... |
| 19:40.33 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:41.57 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 19:46.59 | Shaun2222 | asterwiki: i dont know why the directory path would change from the regular context to the gosub one.. |
| 19:47.16 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 19:47.19 | Shaun2222 | even changed playback to background in the gosub just to see if it was a playback problem.. |
| 19:47.25 | Shaun2222 | couldnt hear it. |
| 19:48.24 | asterwiki | Shaun2222: Where are your sound files stored on your system? /var/lib/asterisk/sounds or /var/lib/asterisk/sounds/en ? |
| 19:51.01 | [TK]D-Fender | asterwiki: This is all posthumous... we'll never know wht the issue is. the evidence is thoroghly gone. |
| 19:52.10 | asterwiki | [TK]D-Fender]: correct... |
| 19:55.09 | *** join/#asterisk talntid (n=asd@98.247.117.146) |
| 19:59.37 | *** join/#asterisk PuroOsso (n=PuroOsso@189.74.141.10) |
| 20:07.17 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:08.07 | chocoz_ | im trying to use a softphone to dial an asterisk box that forwards the call to a second asterisk box. however i cant get the middle box to reach the second box |
| 20:08.26 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 20:09.03 | *** join/#asterisk joako (n=st0n3d@opensuse/member/joak0) |
| 20:10.18 | *** join/#asterisk wopsy (n=80475@AToulouse-754-1-1-6.w90-11.abo.wanadoo.fr) |
| 20:10.23 | asterwiki | chocoz_: Are you using sip or iax to connect the boxes... paste the sip/iax profile you are using in a pastebin |
| 20:11.38 | asterwiki | chocoz_:Also, any CLI error output display you get would be useful for troubleshooting, so paste that in pastebin as well.. |
| 20:13.07 | [TK]D-Fender | ~pb |
| 20:13.08 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 20:13.10 | [TK]D-Fender | ^^^^^^ |
| 20:13.30 | *** join/#asterisk Tim_Toady (n=moi@adsl197-36.kln.forthnet.gr) |
| 20:13.51 | chocoz_ | trying to use sip to connect. http://pastebin.com/m56ad8d62 |
| 20:15.42 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:16.02 | ariel_ | afternoon, |
| 20:16.58 | chocoz_ | typo in the pastebin towards the end, its supposed to be exten 200 for the musiconhold for the 2nd server |
| 20:17.10 | TJNII | chocoz_: And how are the two * boxen supposed to connect? |
| 20:17.12 | ariel_ | I am trying to port a menu over to 1.6 and I am getting an error that says ResponseTimeout,10 no application ResponseTimeout is found, any sample 1.6 menu's around for IVR's |
| 20:17.24 | PuroOsso | need some help: http://pastebin.com/d2311ec95 |
| 20:17.46 | [TK]D-Fender | ariel_: that was deprecated in ------1.2------- |
| 20:18.11 | ariel_ | [TK]D-Fender: yes but what replaced it? I have a few older 1.09 system I am upgrading |
| 20:18.11 | chocoz_ | well, im using a softphone to dial into box 1 on extension 100 and am expecting it to forward to box 2 on extension 200 |
| 20:18.26 | [TK]D-Fender | ariel_: "core show function TIMEOUT" |
| 20:18.54 | TJNII | chocoz_: Yes, I understand expecting. However, you have apparently given it no means or instructions to do so. |
| 20:18.58 | PuroOsso | someone, please.. |
| 20:19.32 | chocoz_ | i thought Dial(SIP/softphone@192.168.1.5/200) would do that? |
| 20:19.51 | [TK]D-Fender | PuroOsso: You are running FreePBX, what you use for memebers, and how its configus are built needs to be taken up with them. |
| 20:20.05 | TJNII | chocoz_: I take that back, I skimmed your pastebin too fast. |
| 20:20.05 | bmoraca | chocoz_: enable SIP debug on the second box and you'll see why it's not working |
| 20:20.28 | [TK]D-Fender | bmoraca: No, loot at the host <- |
| 20:20.36 | bmoraca | ahh |
| 20:20.39 | bmoraca | yeah |
| 20:20.39 | PuroOsso | [TK]D-Fender, ok |
| 20:20.41 | [TK]D-Fender | look* |
| 20:20.42 | bmoraca | pnt |
| 20:22.06 | [TK]D-Fender | And the formatting is GAH |
| 20:22.07 | bmoraca | should have read the thing more closely...was expecting a proxy authentication failure |
| 20:22.36 | asterwiki | chocoz_: first get both servers to inter-connect and ensure that they are registered against each other, then you can dial/forward calls from one to the other in your dialplan |
| 20:23.42 | [TK]D-Fender | registration is not required, having sane peer entries would be a good idea mind you... |
| 20:24.03 | TJNII | chocoz_: Well, there should be more settings in your sip.conf under general, basic sip settings and whatnot, you should register or define the IPs in the sip peer settings, and from a nomenclature standpoint calling the account for the other * box "softphone" is confusing. |
| 20:24.30 | TJNII | chocoz_: Follow asterwiki's advice. sip set debug is a great tool. |
| 20:24.33 | chocoz_ | ive enabled debugging, not sure what im looking for though |
| 20:25.00 | chocoz_ | what docs should i look at for registering the servers against each other? |
| 20:25.06 | [TK]D-Fender | Follow my advice, all the SIP DEBUG in the universe won't show you packets it isn't generating because the host that is parsed isn't valid... |
| 20:25.41 | [TK]D-Fender | And there is NO registration defined. |
| 20:25.44 | [TK]D-Fender | CRAZY PEOPLE |
| 20:26.49 | [TK]D-Fender | ok, checkout time, BBIAB |
| 20:28.08 | *** join/#asterisk dudanogueira (n=dudanogu@189.59.165.56) |
| 20:29.50 | dudanogueira | Hi All! I have a asterisk running at home, and created a IAX extension. I can register from outside, and even receive a call from-pstn to IAX Extension. On the call, I can hear the caller, but they can't hear me! If I am inside the LAN, asterisk being local, everything works fine! |
| 20:30.16 | shido6 | ~sipnat |
| 20:30.18 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:30.18 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:30.42 | TJNII | dudanogueira: How is your phone talking to *? |
| 20:31.00 | dudanogueira | TJNII: X100P (yes, shame on me!!!) |
| 20:31.04 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 20:31.15 | dudanogueira | I am just learning... |
| 20:31.20 | TJNII | dudanogueira: And your connection to your ITSP is IAX? |
| 20:31.43 | dudanogueira | TJNII: that's right! I tried SIP, but I heard that IAX has a better nat problems resolution |
| 20:31.56 | dudanogueira | even if it's IAX i need to fill sipnat? |
| 20:32.24 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 20:32.32 | TJNII | If you're not using SIP than sipnat doesn't apply... |
| 20:32.42 | dudanogueira | shido6: I use freepbx, and read to put the sip_nat confs under sip_general_custom.conf, is that ok? |
| 20:32.58 | dudanogueira | TJNII: that's what I thought! |
| 20:32.58 | TJNII | ~freepbx |
| 20:32.59 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:33.04 | chocoz_ | ah.. so both boxes need to register with each other? |
| 20:33.53 | TJNII | chocoz_: If your boxes IPs are static and you set up the peer section properly, no. |
| 20:34.27 | TJNII | chocoz_: Confused yet? |
| 20:34.30 | dudanogueira | TJNII: I think that's the problem. My * is behind nat, but under a dmz with all ports openned |
| 20:34.46 | *** join/#asterisk ZX81 (n=Matt_Rid@121.74.12.42) |
| 20:35.22 | dudanogueira | But I am "almost" there. Just need to here what people say on the remote extension |
| 20:35.34 | dudanogueira | Next step is to setup a vpn |
| 20:35.35 | dudanogueira | hehehe |
| 20:37.54 | chocoz_ | yes.. very confused. i probably do not have my peers set up properly |
| 20:38.56 | WHYS | where in the source tree is the Dial function located? |
| 20:39.08 | TJNII | chocoz_: No, you don't. :) |
| 20:39.43 | WHYS | hummmm, must be app_dial. :) |
| 20:42.29 | SuPrSluG | chocoz: what is the output of -> sip show peers |
| 20:42.35 | chocoz_ | can you recommend some good in-depth docs that explains a lot of this |
| 20:43.11 | chocoz_ | for *1: box2/s 192.168.1.5 D 5060 Unmonitored |
| 20:43.26 | chocoz_ | for *2: box1/s 192.168.1.4 D 5060 Unmonitored |
| 20:43.50 | TJNII | chocoz_: Good start. Add qualify=yes to the peer section of both. |
| 20:44.01 | *** join/#asterisk dudanogueira_ (n=dudanogu@189.59.165.56) |
| 20:44.43 | SuPrSluG | they're on the same lan so you won't need to register. |
| 20:44.43 | WHYS | ~book |
| 20:44.44 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:45.31 | Katty | really doesn't like ingram micro. |
| 20:46.20 | thehar | i concur katty |
| 20:46.37 | SuPrSluG | chocoz: now the dialplan -> pastebin it |
| 20:48.43 | chocoz_ | http://pastebin.com/m489db21 |
| 20:50.09 | SuPrSluG | they dont' know how to find the other box in the dial plan |
| 20:51.30 | chocoz_ | how can i add that information? |
| 20:52.28 | SuPrSluG | you need to have server A extens= 1XX and server B 2XX or such. you're on your way, but the dialplan needs some info on how to route the calls between the boxes. just like sending them on a route to the pstn |
| 20:53.23 | chocoz_ | as in the extension it should try to reach on the other box? |
| 20:53.24 | SuPrSluG | it's on the wiki site where you got the sip.conf settings. you need the extensions.conf dialplan. |
| 20:53.32 | SuPrSluG | yes |
| 20:54.01 | chocoz_ | okay, so is it Dial(SIP/User@ip/Extension)? |
| 20:54.28 | *** part/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 20:54.36 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 20:54.45 | SuPrSluG | chocoz: http://pastebin.com/m519a31a6 |
| 20:55.44 | SuPrSluG | no more like Dial(SIP/User@ip) would suffice User being the extension |
| 20:56.46 | chocoz_ | ah okay |
| 20:56.48 | TJNII | SIP/exten@peer is what you want, afaik. |
| 20:56.57 | chocoz_ | yes that works! |
| 20:57.14 | SuPrSluG | gtg |
| 20:57.59 | chocoz_ | awesome, thanks |
| 20:58.03 | SuPrSluG | np |
| 20:59.23 | SuPrSluG | now just add more severs to the mix to scale. |
| 21:01.38 | bmoraca | katty: what's wrong with ingram micro? |
| 21:07.11 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 21:07.21 | *** part/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 21:07.51 | *** part/#asterisk dudanogueira (n=dudanogu@189.59.165.56) |
| 21:08.43 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 21:10.19 | *** join/#asterisk sam555 (n=chatzill@udp153808uds.hawaiiantel.net) |
| 21:11.06 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.69) |
| 21:13.23 | Katty | bmoraca: they've not very friendly. |
| 21:15.49 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:16.16 | *** join/#asterisk xpot-mobile (n=james@66-182-88-85.static.sdyl010.digis.net) |
| 21:18.23 | murdock_ut | I think I saw someone ask this question a while ago, but is there a way for the user to hear a different ring when they are calling someone who is currently on the phone? |
| 21:20.56 | Katty | you could check the call status |
| 21:21.06 | Katty | and depending on whether or not the phone supports it |
| 21:21.11 | Katty | send the phone some sort of sip header information |
| 21:21.20 | murdock_ut | Ok. |
| 21:21.21 | Katty | possibly, anyway |
| 21:21.24 | Katty | i've never tried doing it. |
| 21:21.43 | [TK]D-Fender | Katty: technically you have.... |
| 21:23.44 | *** part/#asterisk theHub (n=theHub@69.177.93.21) |
| 21:25.10 | *** join/#asterisk l2trace999 (n=asd@205.245.6.162) |
| 21:29.05 | Katty | [TK]D-Fender: technically, your mom. |
| 21:38.35 | *** join/#asterisk vicidialnow (n=gardo@121.96.159.182) |
| 21:50.13 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
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| 21:54.41 | *** join/#asterisk tgunr (n=tgunr@polymicro.net) |
| 21:57.16 | gardo_away | is away: Gone away for now |
| 21:58.10 | joako_ | Is anyone using an older version of ekiga softphone w/o issues? |
| 21:59.02 | *** join/#asterisk ZX81 (n=Matt_Rid@202.49.106.158) |
| 22:01.13 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
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| 22:10.21 | *** part/#asterisk shadowalke (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 22:10.44 | *** join/#asterisk propellerhead (n=yogurt2u@host240.190-230-87.telecom.net.ar) |
| 22:14.07 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 22:16.09 | leifmadsen | anyone know how to reset a Cisco 7912? Right now when it powers on, it just has the green and red lights turned on, and nothing on the display |
| 22:16.09 | *** join/#asterisk Jimi_ (n=Jimi@unaffiliated/tuxguy) |
| 22:16.21 | Jimi_ | anyone have a pcmu-8000 sample audio stream? |
| 22:16.38 | *** join/#asterisk thuddwhirr (n=wolthuis@mimezine.com) |
| 22:26.41 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 22:49.15 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
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| 22:52.44 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 22:57.00 | *** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com) |
| 22:59.01 | voxter | so i have a weird bug with trying to forward a voicemail to another user on asterisk 1.2 and prepend a message to it, when the recipient receives it, its just the prepended part looped over and over again. anyone experienced this? |
| 23:00.33 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 23:00.44 | manxpower | voxter: I seem to remember some bug like that |
| 23:03.05 | *** join/#asterisk sgtpepper (n=ncorrare@190.55.23.251) |
| 23:03.13 | sgtpepper | Hello everyone |
| 23:03.28 | sgtpepper | I'm having an issue with the originate api via the manager |
| 23:03.44 | sgtpepper | when I put a file on /var/spool/asterisk/outgoing/ |
| 23:03.52 | sgtpepper | it uses the callerid: for both channels |
| 23:03.57 | sgtpepper | when I do it through the api |
| 23:03.59 | sgtpepper | just for one |
| 23:04.23 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 23:11.05 | *** join/#asterisk jimi_ (n=jimi@unaffiliated/tuxguy) |
| 23:11.30 | jimi_ | anyone have a pcmu-8000 file that i can have? |
| 23:13.55 | *** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com) |
| 23:13.59 | sgtpepper | mm.. you can record one |
| 23:14.42 | sgtpepper | exten => *20,1,Answer |
| 23:14.42 | sgtpepper | exten => *20,2,Wait(2) |
| 23:14.42 | sgtpepper | exten => *20,3,Record(asterisk-recording%d:ulaw) |
| 23:14.42 | sgtpepper | exten => *20,4,Wait(2) |
| 23:14.42 | sgtpepper | exten => *20,5,Playback(${RECORDED_FILE}) |
| 23:14.42 | sgtpepper | exten => *20,6,Wait(2) |
| 23:14.42 | box2 | use audacity |
| 23:14.42 | sgtpepper | exten => *20,7,Hangup |
| 23:14.50 | box2 | ooooooo |
| 23:14.54 | box2 | actually that's easy ad nice |
| 23:14.56 | manxpower | sgtpepper: DO NOT flood the channel |
| 23:15.01 | manxpower | ~pastebin |
| 23:15.02 | infobot | [~pastebin] 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 |
| 23:15.04 | sgtpepper | sorry manxpower |
| 23:15.10 | jimi_ | sgtpepper, i dont have an asterisk machine right now.... part of the diagnostic process |
| 23:15.14 | jimi_ | sgtpepper, can you send me one? |
| 23:16.17 | sgtpepper | I think I have one in spanish |
| 23:16.21 | sgtpepper | if you don't mind |
| 23:16.51 | drmessano | Hey HEY |
| 23:16.58 | drmessano | I wrote an Asterisk Haiku |
| 23:17.03 | drmessano | Ready? |
| 23:17.13 | drmessano | . |
| 23:17.16 | drmessano | . |
| 23:17.25 | drmessano | Sorry, NAT problem.. |
| 23:17.34 | drmessano | :( |
| 23:18.31 | *** join/#asterisk geneticx (n=geneticx@adsl-149-109-236.mia.bellsouth.net) |
| 23:19.15 | jimi_ | sgtpepper, i dont.. can you send it? |
| 23:19.22 | geneticx | hello everyone. |
| 23:19.24 | sgtpepper | sure |
| 23:19.36 | sgtpepper | but give me a few min, I'm wrapping up something |
| 23:20.08 | jimi_ | sgtpepper, now |
| 23:20.09 | jimi_ | :P |
| 23:23.36 | sgtpepper | jimi_: PM me an email address |
| 23:27.15 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 23:27.54 | raden_work | is a CICSCO ASA 5505 going to be adequete for 10 computers and 10 phones ? |
| 23:30.05 | *** join/#asterisk protocols (n=protocol@ip-88-153-199-22.unitymediagroup.de) |
| 23:30.11 | shido6 | hrmm |
| 23:30.14 | shido6 | $300 ish? |
| 23:30.40 | shido6 | will you be using VPN? |
| 23:31.57 | raden_work | YES |
| 23:32.05 | raden_work | 2 or 3 users from home |
| 23:32.10 | shido6 | hrmm |
| 23:32.27 | raden_work | why is that not classified as a router but a firwall ? |
| 23:32.29 | shido6 | im starting to like that price |
| 23:33.03 | shido6 | so what will you use on the other end? |
| 23:33.09 | shido6 | from home |
| 23:33.25 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 23:33.35 | *** join/#asterisk w9sh (n=sph@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 23:33.47 | citywok | has anybody run a tftp server wide open to the net to boot phones? |
| 23:34.35 | raden_work | shido6, there client software |
| 23:34.43 | shido6 | for the phone? |
| 23:34.55 | raden_work | shido6, phone wont be on VPN |
| 23:35.00 | shido6 | raden_work: oh..... |
| 23:35.06 | shido6 | raden_work: well okie dokie |
| 23:35.07 | raden_work | just for other things |
| 23:35.10 | raden_work | all phones LAN |
| 23:35.15 | raden_work | 10 computers on LAN |
| 23:35.32 | shido6 | so I can barge in and listen to u talk about the market and darpa projects.........I just cant get the xls or the pdfs? |
| 23:36.03 | shido6 | chuckles |
| 23:37.10 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 23:37.32 | raden_work | ok.... |
| 23:37.36 | Qwell | citywok: no, and I would highly recommend not doing so |
| 23:37.51 | Qwell | You open yourself up to a lot of liability, as well as security issues |
| 23:41.29 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 23:42.04 | *** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com) |
| 23:42.42 | raden_work | Cisco RV082 any good ? |
| 23:44.13 | carrar | *cough* LinkSys |
| 23:45.08 | *** join/#asterisk davidandgoliath (n=David@S0106001d60d4e488.vn.shawcable.net) |
| 23:45.28 | jimi_ | anyone here familiar with Net::SIP/ |
| 23:45.29 | jimi_ | ? |
| 23:46.08 | citywok | Qwell: how do people manage remote phones? just conifgure them statically? |
| 23:46.16 | Qwell | vpn |
| 23:46.35 | citywok | put a vpn router in front of each WFH home user? |
| 23:47.05 | Qwell | sure |
| 23:47.15 | citywok | most of our work from home users have openvpn routers at their houses |
| 23:48.20 | voxter | ok i found a bug in asterisk 1.2, i dont know if it exists in 1.4 yet |
| 23:48.40 | voxter | it happens when you modify voicemail.conf's format= line |
| 23:48.45 | voxter | prepending messages breaks |
| 23:53.44 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-64-138.kgldga.dsl.dynamic.tds.net) |
| 23:54.17 | manxpower | no, removing codecs from the format= line breaks voicemail |
| 23:54.36 | manxpower | Asterisk can't find any of the messages that were recorded in the format that you removed. |
| 23:55.00 | manxpower | don't worry, voxter, it won't be fixed in 1.2 |
| 23:56.20 | voxter | manxpower: no the issue was that i had for some reason specified "wav|wav" |
| 23:56.35 | voxter | probably because an entry on voip-info said that the first argument was used for one thing, and the second for something else |
| 23:56.45 | voxter | but essentially it was clobbering the file when trying to do it a second time |
| 23:56.55 | voxter | or opening itself twice, or something equally stupid |
| 23:57.02 | voxter | i set it to just "wav" and now its okay. |