| 00:00.15 | clh | anyone know what the licensing situation would be if I was using an adtran channel bank for my analog phones but then wanted to connect to nufone using g.729a .... would i need to pay for the codec per channel? ... if so who would i pay? |
| 00:02.28 | wolfson` | c1h: depends where the transcoding is done. if its in asterisk, you'd need a license per simultanious call. and you can buy them from digium |
| 00:02.30 | AvengerX | Nugget: well... I'm with an ATA-186, already configured it for tfpt retrieval of conf and firmware. |
| 00:02.49 | *** join/#asterisk mitchel (~mitchel@69-169-54-180.anhmca.adelphia.net) |
| 00:03.25 | AvengerX | but now i'm looking for info about which servers I must be running in order to e.g. make two telephones w/ ata-186 (each) speak with each other... |
| 00:04.11 | clh | anyone currentlyusing an adtran 750? what codec does it encode to? or does it just pipe the G.711 through? |
| 00:04.15 | AvengerX | i was installing an gatekeeper soft (gnugk) and then I saw there was nothing listening on port 5060, which is a SIP server that I think that will be w/ asterisk package... |
| 00:04.46 | postel | AvengerX: you need an asterisk box, edit sip.conf, add the port registrations as seen on ip/dev on the ATA, ad the extensions on extensions.conf, and voila, call from one port to the other |
| 00:04.53 | Nugget | how did you determine that nothing was listening on port 5060? |
| 00:05.20 | AvengerX | hmmm |
| 00:05.44 | AvengerX | Nugget: netstat -pan | grep gnupg (i was using just gnupg) |
| 00:05.50 | postel | Nugget: tethereal port 5060 works magic |
| 00:05.51 | Primer | are people still working on zaprtc? |
| 00:06.07 | AvengerX | only then I realised that I needed a 'SIP' server to interoperate with that ATA |
| 00:06.10 | Nugget | postel: I know how, I was asking how AvengerX did it. |
| 00:06.24 | Nugget | AvengerX: yes, I think you've come to the right conclusion. asterisk can do this. |
| 00:06.37 | SexyKen | Did I do this wrong: exten => _XXXX,1,Dial(SIP/${EXTEN},20,tr) |
| 00:06.38 | SexyKen | exten => _XXXX,2,Voicemail(u${EXTEN}) |
| 00:06.38 | SexyKen | exten => _XXXX,3,Hangup |
| 00:06.38 | SexyKen | exten => _XXXX,102,Voicemail(b${EXTEN}) |
| 00:06.38 | SexyKen | exten => _XXXX,103,Hangup |
| 00:06.45 | Nugget | many times people mistakenly "telnet hostname 5060" and then declare that it doesn't work. |
| 00:07.13 | AvengerX | postel/nugget: i need to run a SIP server and a GateKeeper?.. or any other server (excluding the TFTP for conf tranfs) |
| 00:07.25 | *** part/#asterisk rene- (~rene-@201.129.49.130) |
| 00:07.28 | Nugget | I have no idea what "a gatekeeper" is, so I don't know. |
| 00:07.56 | AvengerX | Nugget: have u ever conf'ed a cisco ATA-168 for voip? |
| 00:08.03 | AvengerX | (said 186) |
| 00:08.08 | Nugget | an ata-186 can connect to an asterisk server and you can route calls to and from it. |
| 00:08.16 | AvengerX | hmm |
| 00:08.23 | Nugget | no I have not, but the process is well-documented at http://voip-info.org/ |
| 00:08.30 | postel | AvengerX: no dude, just an * box would do it, see what i said earlier |
| 00:08.48 | postel | AvengerX: i got a couple of them here |
| 00:08.49 | markit | SexyKen: what is the problem with your dialplan? I'm not expert, but looks fine to me |
| 00:09.02 | Nugget | 2+101 is not 102. |
| 00:09.02 | AvengerX | lemme try somethings here |
| 00:09.03 | SexyKen | •markit• It wont transfer to voicemail. |
| 00:09.11 | AvengerX | brb |
| 00:09.20 | Nugget | oh, n/m, I missed that the Dial() wasn't pasted. |
| 00:09.25 | AvengerX | Nugget: its 103 :] |
| 00:09.43 | Nugget | yeah, Ken, looks fine to me too. |
| 00:09.44 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-204-71.client.comcast.net) |
| 00:09.46 | Nugget | what's it doing? |
| 00:09.52 | *** join/#asterisk asuffield (~asuffield@83.216.134.182) |
| 00:10.27 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 00:10.27 | markit | SexyKen: when? if it's busy, if it's unavailable, or both? have you defined coicemails in voicemail.conf? |
| 00:11.34 | Darwin35 | anyone here done dial 100 |
| 00:11.38 | SexyKen | Both -- yes defined voicemails in voicemail.conf |
| 00:11.40 | Darwin35 | in exten |
| 00:13.10 | markit | SexyKen: are you sure that the flow goes there? use NoOp(message) to trace on the CLI... I wasted a lot of hours many times trying to fix part of the dialplan were never reached :( |
| 00:13.18 | AvengerX | Briefly, how will I actually link a voIP network to a real phone network? (e.g. to make calls behind my ATA) |
| 00:14.18 | Darwin35 | broadvoice |
| 00:14.24 | Darwin35 | voicepules |
| 00:14.46 | Darwin35 | pick a provider that provides pots/pstn |
| 00:14.51 | AvengerX | Darwin35: that's piece of hardware or software?.. |
| 00:15.04 | Darwin35 | ata is hardware |
| 00:15.21 | Darwin35 | bre is a provideroadvoic |
| 00:15.31 | Darwin35 | broadvoice is a provider |
| 00:15.38 | AvengerX | Darwin35: actually, I'm thinkin in being the provider... so my server will connect ATAs from other cities with telephones here... i think... |
| 00:15.55 | SexyKen | Damn, I must have it configured all wrong. |
| 00:17.16 | Darwin35 | AvengerX where are you and make sure you know what your doing in full before jumping into being a provider |
| 00:17.32 | SexyKen | The problem seem sto be the way I layed out my dial plan, which kinda sucks. :-\ |
| 00:17.44 | Darwin35 | you have to know t1/e1 and setting up did/dod's |
| 00:18.40 | AvengerX | Darwin35: well, for first I will try to use just one conventional telephone line to make up things... |
| 00:19.03 | AvengerX | But the ISP have a E1 line for dialup that should be directed to this job... |
| 00:19.33 | AvengerX | I think I will have to learn a bit about did and dod's :) |
| 00:21.04 | AvengerX | for instance, I am wondering as simple as possible... perhaps to link my home machine w/ an ATA making calls through my home phoneline to the city... and later, I think I could make tests with the e1 line... |
| 00:21.49 | AvengerX | oh.. here goes darwin... :/ |
| 00:22.33 | MrEntropy | if i have 2 X100P cards in my box, the syntax for zaptel.conf is "fxsks=1-2"? |
| 00:23.04 | AvengerX | perhaps 1,2? |
| 00:23.10 | AvengerX | try some :) |
| 00:23.17 | ZX81 | 1-2 is fine |
| 00:23.31 | MrEntropy | cool, thank you |
| 00:23.44 | ZX81 | i have 1-24 |
| 00:23.46 | ZX81 | :-) |
| 00:23.56 | ZX81 | would hate to have to put 1,2,3,4,5,6,7... |
| 00:24.01 | twisted[work] | hehe |
| 00:24.01 | ZX81 | :-) |
| 00:24.17 | markit | SexyKen: btw, if you want to get my attention, prefix your writing with my nick, so I have a notify, while I'm reading another channel ;) |
| 00:24.21 | twisted[work] | ZX81, try 1-24,48-72,73-77 |
| 00:24.28 | ZX81 | twisted[work]: heh |
| 00:24.33 | ZX81 | bastard |
| 00:24.35 | ZX81 | :-) |
| 00:24.39 | twisted[work] | that's on a customer's box |
| 00:24.40 | twisted[work] | not mine |
| 00:24.42 | ZX81 | oh |
| 00:24.43 | ZX81 | :-) |
| 00:24.50 | ZX81 | well this is my home pc :-) |
| 00:24.54 | twisted[work] | my home pc |
| 00:24.55 | twisted[work] | he |
| 00:24.57 | twisted[work] | er heh |
| 00:24.58 | ZX81 | hehe |
| 00:25.20 | twisted[work] | fxoks=1-24 |
| 00:25.32 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-204-71.client.comcast.net) |
| 00:25.34 | twisted[work] | dchan=25-47,49-71 |
| 00:25.38 | ZX81 | heh |
| 00:25.44 | twisted[work] | bchan=48,72 |
| 00:25.55 | ZX81 | :-) |
| 00:26.01 | twisted[work] | or somethin |
| 00:26.06 | ZX81 | :-) |
| 00:26.10 | twisted[work] | heh. |
| 00:26.11 | *** join/#asterisk viperOne (~bulk@82-68-215-6.dsl.in-addr.zen.co.uk) |
| 00:29.04 | AvengerX | wow, asterix seems complete, with that mp3s and .gsm files into it :) |
| 00:29.15 | twisted[work] | wtf is asterix? |
| 00:29.24 | ZX81 | obelix's friend |
| 00:29.25 | AvengerX | asterisk! |
| 00:29.25 | AvengerX | sorry |
| 00:29.30 | twisted[work] | hehe |
| 00:29.30 | AvengerX | LOL |
| 00:29.30 | ZX81 | :-) |
| 00:29.53 | Darwin35 | asterisk is easy |
| 00:30.00 | Darwin35 | once you learn the conf files |
| 00:30.03 | AvengerX | i am just wondering how will I put it into a 30meg free hard disk... :> |
| 00:30.15 | AvengerX | Darwin35: ah u back! :> |
| 00:30.16 | twisted[work] | very carefully |
| 00:30.18 | *** part/#asterisk brettnem (~Brett@216-60-162-174.ded.swbell.net) |
| 00:30.20 | AvengerX | i'm messing with sip.conf now |
| 00:30.27 | dochench | LVM2. ;) |
| 00:31.53 | Darwin35 | sip.coonf is easy extensions.conf is where all the work is done |
| 00:32.07 | twisted[work] | Darwin35, you ain't lyin |
| 00:32.22 | AvengerX | how will I know if my ata-186 uses a-law, u-law, g723.1 or gt729?.. or maybe I allow everything? :) |
| 00:33.16 | *** join/#asterisk enzo123 (~enzo123@24.211.181.116) |
| 00:33.24 | enzo123 | sup all |
| 00:33.37 | AvengerX | the session [grandstream1] in sip.conf must be the LoginID0/1 in my ATA? |
| 00:33.55 | AvengerX | e.g. [mylogin00] or [mylogin01] |
| 00:34.11 | enzo123 | any one here setup a h323 trunk between CCM and * ? |
| 00:34.16 | AvengerX | or I have only to identify it into fromuser=? |
| 00:34.29 | twisted[work] | ROFLMAO |
| 00:34.33 | twisted[work] | today's fortune cookie: |
| 00:34.50 | twisted[work] | "You have an unusual equipment for success, use it properly." |
| 00:35.05 | Darwin35 | lol |
| 00:35.06 | dochench | =) |
| 00:35.14 | Darwin35 | shower time brb |
| 00:35.18 | twisted[work] | tack an "in bed" on to the end of THAT sucker. |
| 00:35.27 | dan2 | twisted[work]: wow, its more realistic, chinese guys who can't speak proper english |
| 00:35.28 | dan2 | :) |
| 00:35.29 | enzo123 | thats what i was going to say |
| 00:36.00 | twisted[work] | hehe |
| 00:36.04 | twisted[work] | this one's getting framed. |
| 00:36.10 | enzo123 | anyone here kick around CCM and * ? |
| 00:36.39 | dan2 | enzo123: are you planning to ask again? |
| 00:36.54 | enzo123 | nope 8> |
| 00:36.57 | m-00kie | gah i hate compiling speech_tools |
| 00:37.08 | *** join/#asterisk asuffield_ (~asuffield@83.216.134.182) |
| 00:37.10 | m-00kie | my kernel compiles faster than that ;/ |
| 00:37.31 | dan2 | asuffield: hey |
| 00:37.33 | crash3m | can anyone point me to a list of commands available in asterisk and their 'help' ? |
| 00:37.35 | bkw_ | where is boris |
| 00:37.48 | bkw_ | show application blah |
| 00:37.49 | bkw_ | help |
| 00:37.52 | bkw_ | show apllications |
| 00:38.10 | dochench | m-00kie: i just finished up that a while back.. has issues doing shared lib, static was fine.. decided festival sounded like crap.. downloading linux frank from cepstral right now. |
| 00:38.12 | enzo123 | the tab key works well too |
| 00:38.14 | dan2 | wow |
| 00:38.22 | dan2 | something is wrong with my irc client |
| 00:38.27 | dan2 | bkw_: hey |
| 00:38.30 | bkw_ | yes |
| 00:38.31 | bkw_ | whats up |
| 00:38.58 | twisted[work] | bkw |
| 00:39.00 | twisted[work] | did you see my fortune? |
| 00:39.10 | bkw_ | twisted[work] what was it? |
| 00:39.15 | enzo123 | twisted scan that sucker |
| 00:39.16 | twisted[work] | "You have an unusual equipment for success, use it properly." |
| 00:39.21 | bkw_ | HAHAHAHA |
| 00:39.26 | bkw_ | you're kidding me |
| 00:39.29 | twisted[work] | enzo123, i just laminated it |
| 00:39.41 | bkw_ | "You have an unusual equipment for success, use it properly." in bed. |
| 00:39.44 | twisted[work] | yes |
| 00:39.47 | enzo123 | drop it in the scanner and post it on the net |
| 00:39.50 | crash3m | asterisk is being run with vvvr and there is a ton of traffic, its not easy to keep a term 'scrolled' to that spot |
| 00:39.58 | twisted[work] | enzo123, it's not that easy |
| 00:40.03 | crash3m | and besides, I was hoping to find something I could easily print, or view with less/more |
| 00:40.04 | twisted[work] | don't have a scanner |
| 00:40.26 | twisted[work] | i could fax it to you :P] |
| 00:40.40 | enzo123 | twisted.. oh.. i have one of those all in one deals.. |
| 00:41.30 | twisted[work] | i have a printer/fax/scanner thingy from hp |
| 00:41.39 | twisted[work] | but i can't use the scanner, as it's hooked up to a print server |
| 00:41.48 | Darwin35 | anyone here done dial 8 dial 100 |
| 00:41.54 | twisted[work] | wait |
| 00:41.57 | twisted[work] | i could fax it to myself. |
| 00:41.58 | twisted[work] | brb |
| 00:42.01 | enzo123 | LMAO |
| 00:42.05 | crash3m | http://www.voip-info.org/wiki-Asterisk+CLI answers my own question, thanks |
| 00:42.09 | Darwin35 | I dont find it in the wiki |
| 00:43.21 | AvengerX | what should I use in extensions.conf if I have no dialplan yet? |
| 00:43.30 | AvengerX | (i want just to barely test the ata with * |
| 00:43.32 | AvengerX | ) |
| 00:43.39 | twisted[work] | okay |
| 00:43.43 | twisted[work] | i'm faxing it to myself in pdf |
| 00:44.28 | enzo123 | allow myself to fax to myself |
| 00:44.29 | *** join/#asterisk bobman (~bobman@falcon.psouth.net) |
| 00:44.36 | twisted[work] | enzo123, want me to email you the pdf? |
| 00:44.43 | enzo123 | sure |
| 00:44.49 | twisted[work] | addy? |
| 00:44.54 | enzo123 | jason@voiceflux.com |
| 00:44.59 | AvengerX | Unable to connect to remote asterisk |
| 00:45.02 | AvengerX | :> |
| 00:45.10 | m-00kie | dont you need the TTS engine itself to use the voices, dochench? |
| 00:45.32 | twisted[work] | enzo123, sent : |
| 00:45.34 | twisted[work] | er :) |
| 00:45.34 | AvengerX | I don't think I want to connect to a remote * |
| 00:45.34 | m-00kie | I couldnt for the life of me find the Cepstral TTS engine o.O all I found was the SAPI5 |
| 00:45.37 | twisted[work] | i sent it to bkw too |
| 00:48.07 | *** part/#asterisk eKo1 (~abc@63.245.57.70) |
| 00:50.51 | dochench | http://www.cepstral.com/cgi-bin/downloads |
| 00:50.58 | enzo123 | twisted ive got it on the web if you want the link |
| 00:51.04 | enzo123 | i just droped it out of pine |
| 00:51.14 | dochench | I just download the linux fank package an installed it to /opt/swift... everything seems to work fine from the shell. |
| 00:51.42 | twisted[work] | enzo123, heh |
| 00:51.43 | twisted[work] | k |
| 00:51.52 | enzo123 | http://www.voiceflux.com/funny.pdf |
| 00:52.06 | *** topic/#asterisk by twisted[work] -> Asterisk: The Open Source PBX || DUNDi (http://www.dundi.com) #dundi || Asterisk/Zaptel/libpri 1.0.3 #asterisk-stable || twisted's forutune: http://www.voiceflux.com/funny.pdf |
| 00:52.21 | enzo123 | haha |
| 00:52.37 | enzo123 | you should have put in bed under it |
| 00:52.42 | twisted[work] | hahah |
| 00:52.52 | twisted[work] | with a marker |
| 00:53.13 | AvengerX | forutune :> |
| 00:53.22 | *** topic/#asterisk by twisted[work] -> Asterisk: The Open Source PBX || DUNDi (http://www.dundi.com) #dundi || Asterisk/Zaptel/libpri 1.0.3 #asterisk-stable || twisted's fortune: http://www.voiceflux.com/funny.pdf |
| 00:53.25 | twisted[work] | thanx |
| 00:53.26 | AvengerX | lol |
| 00:53.33 | AvengerX | t'was nothin |
| 00:53.47 | bkw_ | haha |
| 00:53.56 | almaw | Anyone know much about music on hold royalties and things? It all works, and now I'd like to point it at a Shoutcast server, but I'm wondering if I can do that without the RIAA/BMI/BPI sueing my ass. I'm in the UK. |
| 00:53.58 | AvengerX | Oh my, we need a lot of things working in order to make a VOIP w/ * |
| 00:53.58 | bkw_ | marks fortune beats yours twisted |
| 00:54.04 | twisted[work] | bkw_, which one? |
| 00:54.16 | bkw_ | the able man one |
| 00:54.19 | almaw | I'd really like to use Digitally Imported's trance stream, I guess. :) |
| 00:54.53 | *** join/#asterisk Ahewes (~rsb@67.113.12.162) |
| 00:55.15 | AvengerX | almaw: thats interesting |
| 00:56.29 | AvengerX | i run a shoutcast server, just for fun, but that would be coolest hold music, with a radio on it :) |
| 00:56.36 | twisted[work] | i do that |
| 00:56.40 | twisted[work] | except i use classical streams |
| 00:56.55 | twisted[work] | because most classical music isn't held under copyright law |
| 00:57.06 | AvengerX | * supports streams instead of those mp3's and gsm's inside the source? |
| 00:57.08 | AvengerX | kewl :) |
| 00:57.23 | twisted[work] | ONLY for MOH and mp3player() |
| 00:57.27 | blitzrage | twisted[work]: good idea... I might do that too |
| 00:57.30 | twisted[work] | you can't stream from playback() or background9) |
| 00:57.31 | twisted[work] | er () |
| 00:57.45 | blitzrage | twisted[work]: no wonder your code is so buggy |
| 00:57.49 | almaw | <twisted[work]> because most classical music isn't held under copyright law << That's just not right. |
| 00:58.08 | almaw | It's the performance that's copyrighted. |
| 00:58.08 | bkw_ | almaw you're less likey to get ass fucked if you use classical |
| 00:58.14 | bkw_ | so true |
| 00:58.16 | bkw_ | soo sooo true |
| 00:58.37 | AvengerX | lol |
| 00:58.49 | bkw_ | well we could put britney spears on ours |
| 00:58.56 | bkw_ | lalala |
| 00:59.06 | bkw_ | enzo123 me too... |
| 00:59.13 | bkw_ | oh wait..nm |
| 00:59.17 | bkw_ | NEXT!!! |
| 00:59.18 | almaw | Things like Digitally Imported are odd, because they're all mixes of things. |
| 00:59.25 | almaw | So I'm not sure how that works. |
| 00:59.48 | almaw | They're mostly not artists covered by some studio or other. |
| 01:00.06 | enzo123 | its techo mixes right ? |
| 01:00.23 | almaw | Needless to say, the BPI's web site is full of cruft about piracy, but doesn't actually have any useful info on whether I need to pay them money and how much. |
| 01:00.25 | almaw | Yeah. |
| 01:00.29 | enzo123 | just get a few cd's and rip the mp's |
| 01:00.30 | almaw | They also have a classical channel. |
| 01:00.40 | enzo123 | mp3's |
| 01:01.01 | almaw | The issue is that if I do that, I almost *certainly* have to pay performing rights stuff. |
| 01:01.01 | enzo123 | i hope your not going to have someone on hold for 3 hrs |
| 01:01.11 | almaw | Well, no. |
| 01:01.20 | enzo123 | ah.. got ya |
| 01:01.21 | almaw | It's so hard to find information about this on the web. |
| 01:01.22 | enzo123 | NM |
| 01:01.36 | almaw | It's all a load of bollocks and nothing actually useful. |
| 01:01.48 | bkw_ | almaw you know I think its all bullshit about copyrights and patents |
| 01:01.56 | *** part/#asterisk nem (~nemisis@66.220.23.4) |
| 01:02.01 | bkw_ | I could understand some time to copyright something |
| 01:02.07 | enzo123 | does bollocks = bullshit ? |
| 01:02.14 | almaw | Yeah |
| 01:02.14 | bkw_ | ya |
| 01:02.17 | almaw | Pretty much. :) |
| 01:02.24 | bkw_ | but extending copyrights |
| 01:02.27 | bkw_ | man thats crazy |
| 01:02.27 | paulc | bollocks is so much more classy than bullshit though ;-) |
| 01:02.38 | AvengerX | lol |
| 01:02.40 | bkw_ | its less daft |
| 01:03.05 | almaw | Whether it's bollocks or not, I don't want to have the BPI come storming over and demand £5000 for every trance track we ever stream through the system. |
| 01:03.29 | almaw | It's *so* sexy being able to do MoH from Digitally Imported. And we have exactly the type of customer base who'd appreciate it. |
| 01:03.36 | almaw | I've e-mailed DI about it. |
| 01:03.53 | bkw_ | anyway I can buy DI tracks? |
| 01:03.54 | AvengerX | that's pretty good for newbas: http://www.digium.com/downloads/quick_install_zaptel_asterisk.pdf |
| 01:03.57 | almaw | I don't object to forking out a few dollars a month for their bandwidth if that's what they want. |
| 01:03.57 | AvengerX | =] |
| 01:04.12 | almaw | But it's the BPI bollarks that I'm worried about. |
| 01:04.14 | enzo123 | i think the stream would jack up my connection on cable.. never tried tho |
| 01:05.05 | AvengerX | enzo123: u may do a shoutcast relay into yer internal net to save your cable connection... |
| 01:05.49 | AvengerX | (so everyone that listens to the shoutcast will download music from your local server which used a single stream to bring in the sound to your network) |
| 01:06.14 | enzo123 | kinda like a trunk |
| 01:06.25 | almaw | Bear in mind you only need a 24kbps stream for MoH. |
| 01:07.01 | AvengerX | about installation of *, that zaptel thing I will need if I got any hardware from digium?.. |
| 01:07.03 | enzo123 | ah.. was thinking it was 192k or whatever is on sc |
| 01:07.05 | *** join/#asterisk nem (~nemisis@66.220.23.4) |
| 01:07.10 | *** join/#asterisk cced (~oncemore@222.33.36.198) |
| 01:07.46 | almaw | Well, there are a variety of streams available. |
| 01:08.01 | almaw | I personally think it's crazy to have to pay royalties for doing this. |
| 01:08.12 | almaw | It's on a telephone system at 8kHz ffs. |
| 01:08.51 | almaw | Besides, if you write a bit of software to play internet radio streams, you don't have to pay royalties for each of your users. |
| 01:09.01 | dochench | almaw: there is a bunch of royalty free muzak floating around out there. |
| 01:09.13 | almaw | So if you make a system where you dial in over the phone and it does the same thing, how is that any different? |
| 01:09.37 | *** join/#asterisk dagrim (0@12-218-72-56.client.mchsi.com) |
| 01:09.40 | almaw | How is it any different than me being a downstream internet relayer? |
| 01:10.10 | almaw | I know there's a bunch of muzak. But I want something nice, not something muzak. |
| 01:10.15 | enzo123 | maybe because your playing it to many people at the same time ?? but that would be the same as a party |
| 01:10.26 | almaw | Better off just having intermittent beeps than muzak. |
| 01:11.00 | dagrim | I wish someone would make a you.asterisk.co free dns redirection thing.. lol |
| 01:11.17 | SexyKen | Hrm |
| 01:11.17 | SexyKen | hecking how to run the C++ preprocessor... /lib/cpp |
| 01:11.17 | SexyKen | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 01:11.18 | SexyKen | Any ideas? |
| 01:12.07 | AvengerX | SexyKen: u running as root? |
| 01:12.14 | blitzrage | almaw: I really like Groove Salad. Probably pretty good for MOH too. |
| 01:12.15 | AvengerX | SexyKen: this is happening while compiling *? |
| 01:12.16 | SexyKen | Naturally. |
| 01:12.17 | SexyKen | ;-) |
| 01:12.21 | almaw | You prolly need a more recent version of gcc |
| 01:12.24 | SexyKen | spandsp |
| 01:12.41 | blitzrage | isn't asterisk written in C? Why C++? |
| 01:12.51 | almaw | blitzrage: yeah, but even more dodgy than DI for royalties due to them actually playing real songs. :) |
| 01:13.24 | AvengerX | SexyKen: u truly compiling asterix? |
| 01:13.33 | SexyKen | •AvengerX• Yes. |
| 01:14.04 | AvengerX | there's no C++ preprocessor message around the sources... |
| 01:14.22 | AvengerX | why... are u running configure script?.. |
| 01:14.53 | dagrim | Any know of A working linux phone? lol |
| 01:15.00 | AvengerX | wouldn't you do best a rm -rf, re-unpack it, edit Makefile (if you want to change INSTALL_PREFIX/PREFIX var), and then just 'make install'? |
| 01:15.08 | IOscanner | Anyone has issues using G729 over IAX link between two * boxes? |
| 01:15.17 | IOscanner | I get requested/capability 0x2/0x2 incompatible with our capability 0x10 |
| 01:15.24 | AvengerX | dagrim: working linux phone? mean phone software?.. |
| 01:15.32 | dagrim | AvengerX: yea |
| 01:15.35 | dagrim | my bad |
| 01:15.36 | postel | im so f*cking bored waiting for oracle to mirror the data, are there any public/coffee-house-like voip *rooms* anywhere? |
| 01:16.25 | AvengerX | dagrim: try ohphone (www.openh323.org) |
| 01:16.30 | *** join/#asterisk ChulJin1 (~chuljin@65.211.236.166) |
| 01:16.42 | dagrim | I need a softphone that will compile and run on slackware.. AND actually be able to use my soundcard.. =\ |
| 01:16.48 | dagrim | oh ok thanks ill check it out |
| 01:17.00 | AvengerX | check it... it seems to work fine. |
| 01:17.08 | dagrim | awesome.. =) |
| 01:17.12 | blitzrage | IOscanner: sounds like the other end is only accepting 0x2 codec (gsm?), but you are requesting whatever combo 0x10 gives you (which I presume doesn't include 0x2) |
| 01:17.18 | cced | kphone |
| 01:17.18 | AvengerX | postel: u said u use some ata 186's ya? |
| 01:17.31 | postel | sure |
| 01:17.48 | blitzrage | almaw: true... but I'm not worried about the RIAA :) |
| 01:18.05 | AvengerX | postel: when u installed your *, u used an provider, and connected your * to it?.. |
| 01:18.13 | blitzrage | almaw: but I might be if I was running Asterisk commcially. |
| 01:18.26 | AvengerX | i'm trying to just put * to work and log my ata186 on it... |
| 01:18.33 | IOscanner | I have disallow=all and allow-g729 on both sides in the iax.conf |
| 01:18.34 | almaw | How useless. |
| 01:18.44 | AvengerX | later I see that real world calling... |
| 01:18.51 | postel | AvengerX: yeap, you dont need a provider |
| 01:19.05 | almaw | Having scoured the Performing Rights Society web site, I eventually find "music on hold" under the "clubs" section. You know, nightclubs and things. |
| 01:19.17 | almaw | Muppets. |
| 01:19.34 | AvengerX | postel: when I just compiled it, edited sip.conf and extensions.conf (uh... confunsingly edited), asterisk said: 'could not connect to asterisk server' |
| 01:19.42 | postel | AvengerX: point the ata to asterisk |
| 01:20.03 | AvengerX | postel: i need * working before I point the ata to it, i guess. |
| 01:20.17 | postel | AvengerX: well, makes sense, doesnt it? |
| 01:20.36 | AvengerX | postel: :) so... why is my * trying to connect to another *? ;> |
| 01:20.44 | Atacomm | OUCH: Sipura SPA-841 delayed another month! |
| 01:20.55 | Atacomm | official word, just got it from Sipura |
| 01:21.15 | bobman | Oh crap. |
| 01:21.36 | bobman | Any good reason for the hold up? |
| 01:21.48 | bkw_ | they are trying to hold back so they can meet demand maybe? |
| 01:22.11 | almaw | Right - £142.19 I think is the price. |
| 01:22.32 | Atacomm | probably manufacturing, they didnt say |
| 01:22.44 | almaw | But if we do that, we can use any music we like. Probably. |
| 01:23.17 | ZX81 | bkw_: or trying to build interest so that they sell like hotcakes when they come out |
| 01:23.31 | bkw_ | ZX81 my thinking |
| 01:23.34 | bkw_ | people already want them |
| 01:23.46 | ManxPower | The press release said US$100 - US$120, so that's like GBP 50 |
| 01:24.25 | AvengerX | postel: * keeps saying: 'unable to connect to remote asterisk' |
| 01:24.54 | postel | AvengerX: pastebin the log |
| 01:26.24 | AvengerX | pastebin? |
| 01:26.26 | AvengerX | sorry? |
| 01:26.33 | postel | AvengerX: pastebin.ca |
| 01:26.33 | ZX81 | ~pastebin |
| 01:26.34 | jbot | from memory, pastebin is a place to paste all your conf/debugs/logs for other people in the chatroom to view without flooding the channel. We suggest http://pastebin.ca |
| 01:26.36 | postel | pastebin.com |
| 01:26.43 | postel | ZX81: thanks |
| 01:26.59 | enzo123 | now thats funny |
| 01:27.10 | AvengerX | ah lol |
| 01:27.16 | AvengerX | kewl |
| 01:27.16 | enzo123 | paste happen enough that you have a canned response 8> |
| 01:27.18 | AvengerX | w8 |
| 01:28.01 | mishehu | what's this SPA-841 everybody is talking about today? |
| 01:28.13 | bobman | mishehu: Sipura's entry into the SIP phone market. |
| 01:28.19 | AvengerX | damn :> |
| 01:28.22 | bobman | mishehu: Their first non ATA. |
| 01:28.29 | mishehu | hmm. |
| 01:28.31 | AvengerX | ./asterisk without parameters just ran |
| 01:28.31 | AvengerX | :) |
| 01:28.55 | AvengerX | i think that the '-r' param was messing up things |
| 01:29.29 | PatrickDK | -r is only when asterisk is already running |
| 01:29.36 | PatrickDK | and you want a console |
| 01:29.55 | AvengerX | ahh... now I've got the idea :) |
| 01:31.56 | AvengerX | PatrickDK: that console will give me output (debuggin msgs) or is just a console for input commands to the server? |
| 01:33.09 | AvengerX | lol |
| 01:33.39 | AvengerX | i need to move some spams to /dev/null tho :> |
| 01:33.55 | AvengerX | but I gotta get time to do a script for such filtering |
| 01:34.19 | AvengerX | postel: have you got your ata's to log to syslog servers?.. |
| 01:36.06 | AvengerX | oh my.... some 195.130.87.12 trying to log as root on my box... :> |
| 01:38.27 | postel | AvengerX: well no, i got my cisco routers to log traffic to syslog though |
| 01:38.52 | wolfson` | avengerX: spamassassin works wonders under linux |
| 01:39.35 | PatrickDK | spamassassin is the best thing since the wheel |
| 01:39.49 | wolfson` | i push near 100k messages a day through it, and it works wonders |
| 01:40.15 | PatrickDK | heh, I'm only doing 10k |
| 01:41.18 | AvengerX | postel: i don't know why... but my ata doesnt log things to my syslog (syslogd -r and *.* to /dev/tty12) |
| 01:41.45 | AvengerX | the weirdest thing I realised, is that even without a SIP, my ata gives dialtone... :) |
| 01:41.52 | *** part/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 01:42.02 | ManxPower | AvengerX: Cisco ATA? |
| 01:42.19 | AvengerX | wolfson`: i use spamassassin, now I have to put a filter to actually remove the emails... i'll try the vpopmail/qmailadmin stuff... |
| 01:42.26 | AvengerX | ManxPower: yup. 186. |
| 01:42.31 | *** join/#asterisk nic_oz (~ngbmoreau@ppp152-189.lns1.mel2.internode.on.net) |
| 01:42.51 | AvengerX | ManxPower: no output to my syslog... :/ |
| 01:43.26 | AvengerX | postel: what, mainly, I have to change in extensions.conf in order to test my ata? :] |
| 01:43.45 | *** part/#asterisk bobman (~bobman@falcon.psouth.net) |
| 01:44.13 | AvengerX | also... it is OK to my ATA give dialtone even without the SIP available? |
| 01:44.32 | ManxPower | AvengerX: I think that's normal, but I'm not sure |
| 01:45.04 | PatrickDK | hmm, that is a configurable option |
| 01:46.01 | nic_oz | hi, anyone up to help a newbie ? |
| 01:46.05 | AvengerX | PatrickDK: that should be into my .conf file... |
| 01:46.12 | postel | AvengerX: well, start with sip.conf first |
| 01:46.27 | postel | AvengerX: then ip/ata (put same info there) |
| 01:46.35 | postel | and THEN junp to extensions.conf |
| 01:46.46 | postel | you nedd the ports REGISTERED first |
| 01:46.50 | postel | one step at the time.. |
| 01:46.55 | postel | s/nedd/need |
| 01:47.06 | AvengerX | which ports u saying? |
| 01:47.17 | postel | the nalog ports on your ata |
| 01:47.23 | postel | s/nalog/analog |
| 01:47.23 | AvengerX | ah ok! :) |
| 01:47.27 | PatrickDK | hmm, dialtone is not asterisk controlled |
| 01:47.37 | PatrickDK | so it has nothing to do with sip.conf |
| 01:47.55 | *** join/#asterisk ariel_ (~Ariel@ssf-office.corp.race.com) |
| 01:47.59 | AvengerX | PatrickDK: i saw a directive there to 'avoid ending the dialtone' on '9' keypress.... i think... :) |
| 01:48.29 | PatrickDK | heh, that is for zaptel only |
| 01:48.44 | ariel_ | hello everyone |
| 01:48.48 | AvengerX | drat... the * I just compiled in one slackware did not work into another (which doesnt have space for kernel/includes/headers) |
| 01:48.48 | wolfson` | avengerX: yah, thats exactly what we do, its just a commercial e-mail package that does the actual filtering based on if the sa headers are there. If you are pushing any amount of traffic in 1k's per day, look into spamd/spamc. It will help A LOT. |
| 01:49.51 | AvengerX | wolfson`: i use spam* just to mark the mails as spam or not (i don't think they do further, as stated by spamass'team) but 'tis another subject.. :) |
| 01:50.54 | nic_oz | does SIP transfer work in 1.0.2 ? |
| 01:51.45 | AvengerX | nic_oz: in what 1.0.2? |
| 01:51.48 | wolfson` | avengerX: feel free to /msg me if you have any problems with doing it. |
| 01:52.07 | ManxPower | nic_oz: Yes, but not all types. Use 1.0.3 |
| 01:52.25 | file[laptop] | LLAMA |
| 01:52.43 | twisted | LLAMA! |
| 01:53.07 | nic_oz | AvengerX: asterisk 1.0.2 |
| 01:53.20 | AvengerX | ah ok |
| 01:53.32 | nic_oz | ManxPower: that would be in the CVS I gather ? |
| 01:53.50 | AvengerX | postel: i got my ata already conf'ed... (also, 'tis under NAT)... but I dunno what might be changed in that extensions file... :/ |
| 01:54.33 | ManxPower | nic_oz: join #asterisk-stable and look at the /topic |
| 01:54.43 | AvengerX | postel: it looks like I will need that to connect asterix either to another SIP or to my hardware which will provide 'real' phone lines... |
| 01:55.03 | ManxPower | nic_oz: I assume you saw the 1.0.3 release announcement on the mailing lists? |
| 01:55.04 | AvengerX | (that: that things inside extensions.conf) |
| 01:55.17 | ManxPower | 'cause if you are not on the mailing lists you will go to hell, you know. |
| 01:56.10 | AvengerX | how would I be able to figure out the version of asterisk? Asterisk CVS-HEAD-12/07/04-21:59:24 |
| 01:56.21 | AvengerX | (by issuing asterisk -V) |
| 01:57.02 | IOscanner | Strange I can't get any codecs to work on the IAX2 link other then gsm and ulaw. I try ilbc or g729 and they fail. |
| 01:57.15 | MrEntropy | in order for asterisk to accept calls forwarded from a sip proxy, all i need is a line like "[ser] type=user context=fromser" yes? |
| 01:57.51 | IOscanner | What is this error: requested/capability 0x2/0x2 incompatible with our capability 0x400. |
| 01:57.58 | AvengerX | How I calculate up things for bitmap value like 0x0000... e.g. if there's options 0 1 2 3 4, and I want to use 2 and 4... |
| 01:58.08 | AvengerX | I sum them or I multiply one for the other? |
| 01:58.14 | PatrickDK | just add them |
| 01:58.21 | AvengerX | (0x0006 or 0x0008)? |
| 01:58.27 | PatrickDK | oh, you mean bits 2 and 4? |
| 01:58.47 | AvengerX | yup |
| 01:58.48 | PatrickDK | it would go, 1 2 4 8 16 32 64 ... |
| 01:58.49 | bkw_ | Druken |
| 01:58.52 | bkw_ | doh |
| 01:58.54 | bkw_ | wrong nick |
| 01:58.54 | PatrickDK | don't go 0 1 2 3 4 |
| 01:58.59 | AvengerX | my ATA has some things to mess about that |
| 01:59.12 | IOscanner | both systems are build CVS-v1-0-10/06/04-10:30:21 |
| 01:59.12 | AvengerX | lemme try the sum |
| 01:59.22 | *** join/#asterisk drbyte (~byte@byte.fedora) |
| 01:59.25 | IOscanner | with g729 module added |
| 01:59.32 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc05h.dialup.mindspring.com) |
| 02:00.08 | PatrickDK | that would be 20dec, or 0x0014 |
| 02:01.07 | tzanger | heh |
| 02:01.14 | tzanger | here I'm thinking "20dec = 20th of December" |
| 02:01.26 | tzanger | what kind of bot is jbot? |
| 02:01.42 | AvengerX | http://pastebin.ca/2771 |
| 02:01.58 | AvengerX | i want to use everything of that options |
| 02:02.44 | IOscanner | Well I can get it working with the the cvs-v1-0-11/30/04-00:10:37 version |
| 02:02.49 | IOscanner | strange |
| 02:04.20 | PatrickDK | 0x1071f |
| 02:04.57 | MrEntropy | do i need to "register =>" asterisk to ser in order to receive calls from it? |
| 02:05.15 | JerJer | MrEntropy: depends |
| 02:05.21 | JerJer | if you need to blow thru nat, yes |
| 02:05.26 | AvengerX | PatrickDK: 1071f? |
| 02:05.31 | JerJer | or if you have a dynamic Ip addy |
| 02:05.32 | *** join/#asterisk wolfson (~hehe@65.174.122.198) |
| 02:05.35 | AvengerX | wouldnt that be 56? :> |
| 02:05.49 | ariel_ | anyone here use spandsp? I am having a problem loading asterisk with it. |
| 02:05.52 | PatrickDK | 56? |
| 02:05.58 | dochench | anyone give me a good place to start on using swift/cepstal with asterisk? haven't been able to find much of anything on it in the wiki |
| 02:06.05 | MrEntropy | JerJer: nope, both my asterisk box and the ser box are on public IPs |
| 02:06.06 | JerJer | 0x00150015 |
| 02:06.21 | AvengerX | PatrickDK: you saying about that SyslogCtrl issue? |
| 02:06.29 | PatrickDK | ya |
| 02:06.43 | AvengerX | how'd you came to that value? |
| 02:06.51 | *** join/#asterisk agronqui (~agronqui@207.105.76.114) |
| 02:06.54 | PatrickDK | you take each bit 2^x |
| 02:06.58 | PatrickDK | where x = the bit you want on |
| 02:06.59 | JerJer | MrEntropy: then there is really no requirement to register... just have properly created type=user and type=peer entries |
| 02:07.02 | PatrickDK | then add them all up |
| 02:07.04 | MrEntropy | JerJer: so if i want to only receive calls from ser (ser's forward()) do i just need something along the lines of "[ser] type=user context=fromser"? |
| 02:07.21 | *** join/#asterisk Kumbang (~ecvs@167.205.22.54) |
| 02:07.32 | JerJer | yeah that works |
| 02:07.47 | JerJer | <PROTECTED> |
| 02:08.10 | MrEntropy | JerJer: ok, i'll test around a bit more, having a bit of strife with it =P |
| 02:08.39 | JerJer | asterisk is not trivial to understand |
| 02:09.03 | JerJer | espcially if you plan to leverage some of the more advanced skills |
| 02:09.16 | AvengerX | PatrickDK: err.. that really worked! :] |
| 02:09.20 | AvengerX | now I got messages on my syslog |
| 02:09.53 | ariel_ | ./has/how |
| 02:10.08 | agronqui | MrEntropy: I just finally got ITSP -> ser -> asterisk -> UA working, and I had to have "[ser] type=user context=fromser host=<IPAddress of ser> " |
| 02:10.12 | AvengerX | so you did 2^56?.. oh im such a dumb |
| 02:10.22 | *** part/#asterisk mitchel (~mitchel@69-169-54-180.anhmca.adelphia.net) |
| 02:10.31 | _Simon | modulus_: you there? |
| 02:11.29 | AvengerX | 1071f... the last option is really 16, which is F in hex... but the one before last, wasnt 10? |
| 02:11.37 | AvengerX | (A) |
| 02:11.37 | PatrickDK | 16 !=f |
| 02:11.39 | PatrickDK | 15 = f |
| 02:11.59 | AvengerX | ah truly... omg I messed up even more... :> |
| 02:12.05 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 02:12.14 | PatrickDK | and it's not 2^56 |
| 02:12.16 | dev5 | 0x20 = 16 ;) |
| 02:12.18 | agronqui | MrEntory: and in extensions.conf you need "[fromser] exten => <some pattern that matches the inbound DNIS> + actions" or else it will say no extension found... |
| 02:12.28 | dev5 | ops |
| 02:12.30 | PatrickDK | 0x10 = 16 |
| 02:12.34 | dev5 | yes ;) |
| 02:12.51 | PatrickDK | hmm, to get what I got you do 2^16 + 2^10 + ... |
| 02:12.58 | AvengerX | ok.... 0 1 2 3 4 5 6 7 8 9 a b c d e f 10 11... |
| 02:13.09 | AvengerX | ahhhhhhhhh!!! |
| 02:13.21 | MrEntropy | agronqui: ok, but i got something very similar...and it doesn't work |
| 02:13.46 | MrEntropy | agronqui: however i don't have the host= thing. let me try that |
| 02:14.00 | PatrickDK | heh, I would kill my math teacher, doing my calc in binary, heh |
| 02:14.23 | _Simon | Anyone here familiar with RealTime? I've got some tables created, added a line to extconfig.conf but not sure how to tell iax.conf to load from db, do I have to point it to the line in extconfig.conf somehow? |
| 02:15.07 | PatrickDK | 020 = 16 :) |
| 02:15.46 | MrEntropy | agronqui: could you give me an example of your working ser.conf. at least how you forwarded to asterisk? |
| 02:16.17 | MrEntropy | i just tried with host= =/ no go |
| 02:16.29 | *** join/#asterisk GarryH (~guangyaoc@S0106009027bbc526.ed.shawcable.net) |
| 02:16.36 | AvengerX | PatrickDK: to show that I've understood: 2^0 2^1 2^2 2^3 2^4 2^8 2^9 2^10 2^16 |
| 02:16.36 | AvengerX | 1 2 4 8 16 256 512 1024 65536 = 67359 = 1071f |
| 02:16.39 | AvengerX | :> |
| 02:16.55 | MrEntropy | agronqui: does the phone (which is registered to ser) have to have an account on the asterisk box? i wouldn't think so... |
| 02:17.04 | GarryH | Good evening everyone! Any me could help me with call parking. |
| 02:17.05 | Atacomm | ~seen implicit |
| 02:17.06 | jbot | implicit <~implicit@ip68-5-148-1.oc.oc.cox.net> was last seen on IRC in channel #asterisk, 1d 2h 21m 2s ago, saying: 'eKo1: lol'. |
| 02:17.30 | AvengerX | ~sip |
| 02:17.31 | jbot | X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 02:17.39 | AvengerX | kewl |
| 02:18.02 | cced | why busydetect can work sometimes,then sometimes not. |
| 02:18.34 | agronqui | MrEntropy: sure, lemme pastebin it real quick, ok? |
| 02:19.09 | MrEntropy | agronqui: sure, thanks |
| 02:19.14 | AvengerX | now let's get that ata into the SIP |
| 02:19.23 | AvengerX | (so many abbreviations) |
| 02:19.47 | AvengerX | the ATA into the SIP to call via E1 to the ISP around IP to the hell with it! lol |
| 02:20.00 | GarryH | I have a grandstream 101 with latest firmware. I have some trouble with call parking. Help! |
| 02:20.15 | dan2 | bkw_: ping |
| 02:20.34 | agronqui | MrEntropy: the UA (mine is actually an IVR) does not register with asterisk, but it (for my use at least) has to have it's IP address hardcoded into a channel so that you can do the Dial(SIP/something) in extensions.conf..... |
| 02:21.28 | *** join/#asterisk wisdom (~cgucker@gait.onesc.net) |
| 02:21.52 | MrEntropy | agronqui: i'm sorry, i don't understand |
| 02:22.10 | MrEntropy | agronqui: you mean ser has to have the ip hardcoded? |
| 02:23.23 | AvengerX | wonderfull to see the numbers pop up in the screen as I press them :> |
| 02:23.34 | AvengerX | (in the analog phone) |
| 02:23.50 | AvengerX | Dec 8 00:23:10 WARNING[3655]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 3404604300@10.3.172.205 for seqno 1 (Critical Response) |
| 02:24.00 | AvengerX | i think that now I am under an 'NAT' issue... :/ |
| 02:25.05 | agronqui | MrEntropy: http://pastebin.ca/2772 |
| 02:26.04 | agronqui | MrEntropy: you want to have Asterisk dial an extension when the call comes in, aye? as opposed to doing voicemail in asterisk or somesuch? |
| 02:27.17 | MrEntropy | agronqui: yes, when a call is forwarded from ser to asterisk, asterisk dials it out a zap channel. |
| 02:27.47 | agronqui | MrEntropy: so you have to have the UA IP address either registered (using dynaminc, I haven't tried this...) or hardcoded in a seperate channel section in sip.conf, it's like, the in channel from ser, and the out channel to the UA...... |
| 02:27.50 | AvengerX | now everything seems okay! :D |
| 02:27.50 | AvengerX | Dec 8 00:27:22 WARNING[3655]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 2170678217@200.199.110.111 for seqno 1 (Critical Response) |
| 02:28.10 | agronqui | MrEntropy: oooh, you going out zap channel, I see, that's different from mine, mine is oing back out SIP......long story ;) |
| 02:28.19 | AvengerX | a -HP will make * read every .conf file? |
| 02:28.38 | MrEntropy | agronqui: but i see from your config where i may have gone wrong, i got my if(uri){forward()}; in the wrong place it seems |
| 02:29.24 | mutilator | i still have a job!! |
| 02:29.28 | mutilator | amazing |
| 02:29.31 | agronqui | MrEntropy: oh, that'll do it, I hope :) |
| 02:29.48 | MrEntropy | agronqui: i |
| 02:29.59 | MrEntropy | agronqui: i'll give it a shot now |
| 02:31.13 | agronqui | MrEntropy: and are you using "host*CLI> sip debug" in an asterisk console (asterisk -r) to see what exactly is going on? that helped me tremendously to see where stuff was failing..... |
| 02:31.32 | AvengerX | how do I hear an musiconhold? :> |
| 02:31.45 | PatrickDK | setup an exten for it |
| 02:31.51 | PatrickDK | or put yourself on hold |
| 02:32.02 | AvengerX | there's already the default songs from the package |
| 02:32.09 | AvengerX | so, how I do it? :) |
| 02:32.18 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
| 02:32.18 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:33.48 | dochench | twisted: was wondering if you could point me in the direction for interfacing asterisk with cepstral voices? haven't been ab;e to find diddly, but did find a msg from "twisted" from sept in the users mailing list. |
| 02:33.55 | *** join/#asterisk docelm0 (~me@67.106.194.90.ptr.us.xo.net) |
| 02:37.13 | MrEntropy | agronqui: i have this (after correcting where to do the forward()), asterisk recieves the call...but drops http://pastebin.ca/2773 |
| 02:37.18 | AvengerX | heh |
| 02:37.21 | AvengerX | 1234 :> |
| 02:37.22 | AvengerX | Dec 8 00:36:38 WARNING[22414]: channel.c:1920 ast_request: No channel type registered for 'Console' |
| 02:37.22 | AvengerX | Dec 8 00:36:38 NOTICE[22414]: app_dial.c:803 dial_exec: Unable to create channel of type 'Console' (cause 66) |
| 02:37.39 | AvengerX | what sould I do to fix that? |
| 02:38.01 | cced | why busydetect can work sometimes,then sometimes not. |
| 02:38.10 | cced | hello everyone~ . |
| 02:38.22 | Nugget | it knows when you're just pretending to look busy. |
| 02:38.29 | Nugget | but really you're playing solitaire |
| 02:39.12 | MrEntropy | agronqui: i'm looking at the sip debug on asterisk...it says that it's looking for the number under default, when it's meant to look under fromser. |
| 02:39.40 | agronqui | MrEntropy: did you see my post about sip debug:? turn that on and watch as you call in, see what reason it gives for the error, like for a while I wasn't matching my DNIS to a extension in the context (in extensions.conf) that was specified for the ser channel, so it dropped me....... |
| 02:39.52 | viperOne | hi ! i just installed asterisk, trying to run a simple prepaid app with astcc. my question is: how can i create a prompt tree asking 1.) to choose between english and spanish, then 2.) to enter the PIN number in the language choosen in 1 |
| 02:40.47 | file[laptop] | viperOne: modify astcc. |
| 02:40.59 | MrEntropy | agronqui: asterisk says "Found no matching peer or user for 'ser.box.ip.addr:5060'" ... but there clearly is one in sip.conf =/ |
| 02:41.11 | twisted | i'm about to destroy a 7960 |
| 02:41.21 | twisted | suddenly, it doesn't like the damned .cnf file it's been using all along |
| 02:41.25 | bkw_ | AvengerX load the channel console driver |
| 02:41.34 | file[laptop] | MrEntropy: you need to have host=<IP address of ser box>, have it as a type peer, and set insecure=very |
| 02:41.56 | AvengerX | bkw_: uh?.. is that related to any hardware, like a sound board or something else? |
| 02:42.05 | bkw_ | sound board |
| 02:42.09 | bkw_ | do ya have one? |
| 02:42.44 | MrEntropy | file[laptop]: why peer? i thought user is the one that places calls to asterisk? |
| 02:42.58 | AvengerX | one of that mods around /usr/lib/asterisk/modules#? |
| 02:43.03 | file[laptop] | MrEntropy: yeah but if it's a peer it'll cause asterisk to match via IP address and send it into the correct context |
| 02:43.06 | AvengerX | I need a sound board in the machine running asterisk? |
| 02:43.16 | agronqui | MrEntropy: hmmmmmm.....do you have host=addy or host=ip:port?? I have host=addy |
| 02:43.16 | file[laptop] | MrEntropy: asterisk will match to that entry based on the IP address... doesn't do it on user, not that I've seen atleast |
| 02:43.18 | m-00kie | hrm |
| 02:43.22 | AvengerX | wouldnt it get a stream from my ATA and record it... or play a mp3 to my stream? |
| 02:43.25 | file[laptop] | try both though, never know! |
| 02:43.33 | Rez | with LNP, can a SS7 connected switch assume the routing of any NPNA phone number? |
| 02:43.34 | agronqui | MrEntropy: and I used friend, to be sure, hehe |
| 02:43.38 | AvengerX | bkw_: wouldnt it get a stream from my ATA and record it... or play a mp3 to my stream? |
| 02:43.57 | m-00kie | i havent even finished setting up my phone numbers and already cdr's recording incoming calls to unregistered #'s.. |
| 02:44.00 | m-00kie | hope thats not a bad sign :P |
| 02:44.08 | AvengerX | bkw_: (indeed, the box haves a onboad c-m bundled into it) |
| 02:44.27 | MrEntropy | agronqui: ok but the thing in the square [] brackets...that's just a marker? right? it can be anything you want...nothing to do with authentication? |
| 02:44.38 | *** join/#asterisk GarryH (~guangyaoc@S0106009027bbc526.ed.shawcable.net) |
| 02:44.44 | *** join/#asterisk Sefer (~Ouroboros@200-153-211-190.dsl.telesp.net.br) |
| 02:44.49 | AvengerX | bkw_: there's also some modules into /usr/lib/asterisk/modules... but no 'console-like' mod |
| 02:45.10 | file[laptop] | chan_alsa and chan_oss are console like modules |
| 02:45.11 | Sefer | hi folks! |
| 02:45.23 | agronqui | MrEntropy: the thing in [] in sip.conf (channels) or the thing in [] in extensions.conf (context)?? |
| 02:45.25 | Sefer | i am a newbie about voip |
| 02:45.33 | Sefer | and never looked before at asterisk |
| 02:45.37 | *** join/#asterisk BoRiS (~boris@S01060050da67299b.wp.shawcable.net) |
| 02:45.44 | Sefer | i am just trying to make it now... |
| 02:45.48 | MrEntropy | agronqui: oh, sorry, sip.conf |
| 02:45.48 | file[laptop] | AvengerX: they use ALSA and OSS respectively so that the asterisk box can act as a phone essentially... |
| 02:45.52 | Sefer | and discover some features about it |
| 02:45.58 | AvengerX | bkw_: also, that mods don't seem like kernel's :> |
| 02:46.04 | AvengerX | Sefer: i see... |
| 02:46.07 | GarryH | Can someone help me with call parking? |
| 02:46.07 | Sefer | i will try to set up it here now |
| 02:46.20 | file[laptop] | BoRiS: once a whore... |
| 02:46.30 | Sefer | but can i talk only using a broad band conection through the internet? |
| 02:46.46 | agronqui | MrEntropy: in sip.conf it is just a marker, but the context defined for that channel must match a section in extensions.conf, right? |
| 02:46.50 | AvengerX | Sefer: i think you'd better setup it and come with questions to us to try to help you :) |
| 02:47.10 | MrEntropy | agronqui: yes, i got that much =) |
| 02:47.14 | AvengerX | Sefer: I think that a good 56k modem can handle a single voip connection... not sure. |
| 02:47.25 | Sefer | AvengerX, great! |
| 02:47.37 | AvengerX | Sefer: are u using an cisco ata-186? |
| 02:47.37 | agronqui | MrEntropy: so did you try with type=friend ?? |
| 02:47.38 | AvengerX | :> |
| 02:47.50 | Sefer | AvengerX, no... |
| 02:47.51 | MrEntropy | agronqui: i'm just about to |
| 02:48.01 | Sefer | i am using a parks zyxel prestige |
| 02:48.13 | AvengerX | Sefer: ah... the cisco ata seems great! |
| 02:48.19 | AvengerX | (i'm newbah too) |
| 02:48.36 | Sefer | AvengerX, so we can help each other to reach some goals! |
| 02:48.37 | Sefer | :) |
| 02:48.49 | AvengerX | today's the first day, and I've already installed asterisk and made my ata to (try to) talk with the server |
| 02:48.51 | Sefer | AvengerX, thanks AvengerX i will read the asterisk hand book |
| 02:48.52 | Sefer | :) |
| 02:48.52 | bkw_ | BoRiS yo ho |
| 02:49.24 | AvengerX | Sefer: yea... installation is kinda simple as long as your box has good compiling capabilities... (maybe compabilities) |
| 02:49.37 | MrEntropy | agronqui: hmm, friend works...but now i got a zap problem...i had zap working before...i'll try and figure it out...got a bit of a cabling tangle too...back in a sec. |
| 02:49.43 | bkw_ | BoRiS yo yo yo |
| 02:49.45 | bkw_ | wake up BoRiS |
| 02:49.55 | AvengerX | bkw_: if I load my sound drivers, i'll free up /dev/dsp, but I don't think that there will be /dev/Console/dsp |
| 02:49.57 | twisted | fuck. |
| 02:50.04 | bkw_ | yes? |
| 02:50.07 | bkw_ | whats wrong ? |
| 02:50.09 | AvengerX | bkw_: and... why I need the sound board?.. I don't understand... |
| 02:50.13 | bkw_ | 666 |
| 02:50.14 | bkw_ | baby |
| 02:50.16 | *** join/#asterisk NormAst (NormAst@Ottawa-HSE-ppp4121010.sympatico.ca) |
| 02:50.29 | AvengerX | bkw_: * seems to be looking for Console/dsp |
| 02:50.48 | GarryH | Anyone familiar with call parking? |
| 02:50.50 | BoRiS | ok |
| 02:50.55 | bkw_ | BoRiS 666 |
| 02:51.00 | agronqui | MrEntropy: I gotta roll out of work and go to a presentation.........maybe file would be so nice as to help you out? :) |
| 02:51.01 | AvengerX | GarryH: what is call parking? :> |
| 02:51.23 | agronqui | MrEntropy: and also: I have no hardware, so zaptel configs will be beyond me :) |
| 02:51.29 | file[laptop] | AvengerX: there's an extension that dials the console, the console uses your sound card via chan_alsa or chan_oss that allows asterisk to act as a softphone |
| 02:52.34 | MrEntropy | agronqui: sure, i've done zap before on my own. thanks a lot for the help |
| 02:52.49 | MrEntropy | agronqui: truly appreciated |
| 02:52.51 | agronqui | MrEntropy: np, I been lurking in this channel for days, thought I'd pipe up ;) |
| 02:53.05 | brc_ | HOLA |
| 02:53.09 | GarryH | You can park a call at a specific number, then hangup. pickup another phone, call that number, continue to talk. |
| 02:53.14 | brc_ | BORIS"!!!!!!!!!!!!!!!!!!!!!11111 |
| 02:53.15 | AvengerX | file[laptop]: how to register that console?.. |
| 02:53.20 | BoRiS | brc!!!!!!!!!!!!!!! |
| 02:53.22 | BoRiS | :) |
| 02:53.39 | file[laptop] | AvengerX: you load the chan_oss or chan_alsa module, they use your soundcard |
| 02:53.44 | *** join/#asterisk Hydroxide (user@adsl-98-2.bway.net) |
| 02:53.45 | AvengerX | file[laptop]: also... why I need the sound card... I'm using an ATA with an analog phone here at home... the server will only be used for 'recording a voicemail'... |
| 02:53.59 | mikegrb | you don't need a soundcard |
| 02:54.12 | AvengerX | file[laptop]: i truly need that even in this scenario I told right now? |
| 02:54.14 | file[laptop] | AvengerX: you don't, it's just it's in there in the default configuration |
| 02:54.23 | file[laptop] | AvengerX: you can remove it if you want |
| 02:54.28 | AvengerX | CONSOLE= var? |
| 02:54.35 | Hydroxide | hi all ... I know this is an FAQ, but I can't get it to work. how do I call another context and then return to where I left off in the dialplan after that context completes/returns/hangs up/something? |
| 02:54.37 | file[laptop] | learn about the dialplan and you'll know what I mean |
| 02:55.06 | Hydroxide | I'm trying various incarnations of Dial() with local channels, but the precise correct commands elude me |
| 02:55.22 | AvengerX | file[laptop]: lemme look at it on one wiki :> |
| 02:55.54 | GarryH | file[laptop]: Can you help me? I have some call parking issues. |
| 02:56.30 | Hydroxide | I'm basically trying to do the equivalent of a subroutine call and then returning from that subroutine in most programming languages |
| 02:56.55 | file[laptop] | GarryH: erm I'm departing |
| 02:57.13 | AvengerX | :> |
| 02:57.18 | AvengerX | slumber? |
| 02:57.44 | *** join/#asterisk Yellow_Fuzzy (~yellowfuz@c211-30-2-126.wavrl1.nsw.optusnet.com.au) |
| 02:57.47 | Yellow_Fuzzy | hi |
| 02:58.45 | Hydroxide | I forgot to mention, I'm using asterisk 1.0.2. I see the /topic and just asked in #asterisk-stable as well. |
| 02:59.12 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
| 02:59.12 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:59.18 | AvengerX | Sefer: interesting text 4 ya: http://www.voip-info.org/wiki-Bandwidth+consumption |
| 02:59.59 | AvengerX | Sefer: there's also a link for low speed voIP |
| 03:00.50 | Sefer | cool |
| 03:00.51 | Sefer | :) |
| 03:00.54 | AvengerX | file[laptop]: is there any alias for DialPlans? (no 'DialPlan' found in the wiki...) |
| 03:00.55 | Sefer | thanx! |
| 03:01.10 | Sefer | do you think is it hard to set up? |
| 03:01.10 | file[laptop] | extensions.conf |
| 03:01.15 | AvengerX | Sefer: that's a plenty wiki... |
| 03:01.30 | BoRiS | file[laptop]: *ALWAYS* a whore |
| 03:01.34 | AvengerX | file[laptop]: into the file has some explaining? |
| 03:01.35 | file[laptop] | finally |
| 03:01.46 | file[laptop] | AvengerX: no I mean look it p |
| 03:01.48 | file[laptop] | er up |
| 03:02.04 | AvengerX | i found dialplan into my ata's setup |
| 03:02.11 | AvengerX | boris: try page up :> |
| 03:03.05 | file[laptop] | I meant the asterisk dialplan |
| 03:03.10 | file[laptop] | it tells asterisk how to route/handle calls |
| 03:04.45 | AvengerX | perhaps they're the same... |
| 03:04.45 | file[laptop] | no, they're not |
| 03:04.45 | AvengerX | Then that CONSOLE= thing is about routing a call?.. |
| 03:04.45 | file[laptop] | the CONSOLE= thing is setting a variable |
| 03:04.45 | file[laptop] | if you go read, you'll learn it |
| 03:04.45 | AvengerX | well... it looks a bit weird for me right now... |
| 03:04.51 | file[laptop] | that is why one has to go read and learn |
| 03:05.03 | AvengerX | :) |
| 03:05.26 | *** join/#asterisk iVoice (~iVoice@81-174-11-211.f5.ngi.it) |
| 03:05.42 | AvengerX | ok ok... i won't bother you again with this issue. |
| 03:06.22 | Hydroxide | okay, if nobody can ansewr my question, can someone at least suggest somewhere else to ask? I'm surprised the answer is obscure enough that most of you experienced asterisk users don't know it immediately |
| 03:06.35 | Hydroxide | s/ansewr/answer/ |
| 03:06.40 | AvengerX | Hydroxide: ?? |
| 03:06.56 | file[laptop] | Hydroxide: I don't understand what you want |
| 03:07.05 | AvengerX | Hydroxide: don't be so rude.... much of us are away :> |
| 03:07.29 | Hydroxide | AvengerX: I apologize if I sounded rude ... I'm amazed there's nothing clear about this on the wiki, since it seems like a simple thing |
| 03:07.43 | Hydroxide | basically, I'm trying to call from one context to another and then return to where I left off |
| 03:07.50 | AvengerX | but it seems like a devel issue isnt it? |
| 03:07.54 | Hydroxide | when the destination context finishes |
| 03:08.06 | tzanger | Hydroxide: I do not believe that is possible |
| 03:08.09 | *** join/#asterisk aspworld (~richard@209.91.159.221) |
| 03:08.13 | Hydroxide | the "Asterisk local channels" page on the wiki suggests that it is |
| 03:08.16 | tzanger | you're looking for a gosub |
| 03:08.19 | AvengerX | and uh... i'm just a newbie in *... even in voip |
| 03:08.43 | tzanger | macros might do what you're after but that's about it |
| 03:08.46 | Hydroxide | "chan_local is a pseudo-channel. Use of this channel simply loops calls back into the dialplan in a different context. Useful for recursive routing; it is able to return after to the dialplan after call completion. |
| 03:08.49 | Yellow_Fuzzy | Dec 8 13:57:07 WARNING[2612]: chan_iax2.c:5550 socket_read: Call rejected by 61.29.15.133: No authority found |
| 03:08.50 | Hydroxide | " |
| 03:08.52 | AvengerX | I can't get the voicemail to work.. :/ |
| 03:08.55 | Yellow_Fuzzy | i keep getting that error |
| 03:09.11 | Hydroxide | tzanger: I can't use macros because the subroutine context needs to use multiple different extensions |
| 03:09.12 | dochench | is there a completed 2nd version of the handbook? the latest I can find is a draft from 2003. |
| 03:09.23 | Yellow_Fuzzy | and ive made sure my username and pass are correct |
| 03:09.29 | Moc | hi all |
| 03:09.34 | tzanger | Hydroxide: you can't pass the desired extension to the macro as an argument? |
| 03:09.39 | tih | Hydroxide: that sounds like you want a Dial(Local/......), doesn't it? |
| 03:09.40 | file[laptop] | Hydroxide: Dial(Local/145@default) |
| 03:09.50 | file[laptop] | dials extension 145 in default context |
| 03:10.05 | tzanger | file[laptop]: interesting I did not know this |
| 03:10.14 | Hydroxide | tih, file[laptop]: yes, I have that, but how do I return to the calling context when the called context is "done"? |
| 03:10.18 | file[laptop] | Local is fun |
| 03:10.24 | file[laptop] | Hydroxide: you can't |
| 03:10.27 | tzanger | I'd have done Macro(somemacro,145) |
| 03:10.41 | tzanger | or _XXX,1,Macro(somemacro,${EXTEN}) |
| 03:10.47 | tzanger | _XXX,2,CarryOn() |
| 03:10.50 | Moc | Anyone offer 1800 that work from Canada or 514,418,819,450 dids ? |
| 03:10.50 | file[laptop] | Hydroxide: oh misread that |
| 03:10.51 | Hydroxide | file[laptop]: the "Asterisk local channels" wiki page says "it is able to return after to the dialplan after call completion. |
| 03:10.53 | file[laptop] | sorrrrrrrrrrrrry |
| 03:10.54 | Hydroxide | " |
| 03:10.59 | AvengerX | Yellow_Fuzzy: you just trying to connect to your own * server? |
| 03:11.16 | file[laptop] | Hydroxide: look at the stuff for Dial... there's a thing for continuing in the dialplan when the channel is hung up |
| 03:11.24 | Yellow_Fuzzy | no im trying to connect make a call through my voip provider to the PSTN network |
| 03:11.25 | file[laptop] | so when the Local channel is hung up, it'll go back to the calling context |
| 03:11.33 | tzanger | file[laptop]: that's 'g' flag |
| 03:11.36 | *** join/#asterisk DyOS (~me@ip68-3-226-48.ph.ph.cox.net) |
| 03:11.45 | file[laptop] | unless there's more stuff in the extension you call in Local |
| 03:11.49 | *** join/#asterisk onixx (1000@CPE0040f47145d1-CM000f9f7f2290.cpe.net.cable.rogers.com) |
| 03:11.57 | tzanger | file[laptop]: that only ONLY happens when the REMOTE end hangs up-- if the originator hangs up it does NOT go on (which makes sense, the channel is gone) |
| 03:12.02 | Hydroxide | ah, I looked at those pages but didn't see the "g" flag |
| 03:12.04 | AvengerX | Yellow_Fuzzy: ah... no idea... sorry... that should be account/perms with your provider... |
| 03:12.10 | file[laptop] | tzanger: yeah trying to think |
| 03:12.15 | Hydroxide | tzanger: but the called party would be hanging up ... namely, the subroutine context |
| 03:12.26 | onixx | hi all, anyone here use a cisco 79XX phone in france, that would know if this phone can have french menus ? |
| 03:12.28 | tzanger | so that'd be fine then |
| 03:12.39 | Yellow_Fuzzy | my provider and i have checked the user details a number of times and they are right |
| 03:12.50 | pfn | onixx I think cisco only has ENU supported at present |
| 03:12.59 | Hydroxide | i.e., if context A does Dial(Local/500@B,,g), then it would be B that would Hangup. does that work? (I'll try it while you answer) |
| 03:13.12 | *** join/#asterisk wolfson` (hehe@cpe-68-187-188-219.man.nc.charter.com) |
| 03:13.18 | AvengerX | onixx: I think cisco have some .kup files with translations... but not sure about the voices... |
| 03:13.33 | cced | how to write linux drive ? x101 clone card/ |
| 03:14.00 | onixx | AvengerX: voices ? |
| 03:14.06 | nic_oz | is there anyway enable a debug option that will let me see if asterisk is getting a message when I press the transfer key on my sip phone ? |
| 03:14.07 | AvengerX | onixx: perhaps if you get into cisco's HQ page as a registered client you can get any translation you might want to... |
| 03:14.49 | AvengerX | onixx: didn't you talk about the spoken menus from the cisco gadget? |
| 03:15.06 | AvengerX | (talk/told ns) |
| 03:15.13 | Hydroxide | didn't appear to work :< |
| 03:15.25 | pfn | nic_oz sip debug |
| 03:15.27 | *** join/#asterisk TwoCanChe (ZDEFAULT@i216-58-90-73.avalonworks.net) |
| 03:15.33 | *** part/#asterisk TwoCanChe (ZDEFAULT@i216-58-90-73.avalonworks.net) |
| 03:15.40 | onixx | AvengerX: no... just the menus on the screen... like Answer soft key ... |
| 03:15.47 | *** join/#asterisk TwoCanChe (ZDEFAULT@i216-58-90-73.avalonworks.net) |
| 03:16.21 | AvengerX | onixx: ah ok :> |
| 03:16.36 | pfn | onixx cisco's official word 9as of the last month or so, is that they only have english menus |
| 03:16.58 | AvengerX | onixx: i'm using one ata 186 which haves voice menus (if you hang the phone and press a button in the ata) |
| 03:17.20 | onixx | pfn: that's weird... not only for french... but what about spanish, chinese... ;) |
| 03:17.34 | pfn | onixx read cisco's faq page, they say only english so far |
| 03:18.14 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
| 03:18.14 | *** mode/#asterisk [+o twisted] by ChanServ |
| 03:19.05 | DyOS | can someone tell me how to configure my * server to make outbound calls using an iax peer such as nufone? |
| 03:19.10 | onixx | pfn: thanks for the info, I guess I'll keep looking |
| 03:19.49 | Hydroxide | ok, using a command like Dial(Local/500@newcontext/n,,g) and then calling Hangup from within newcontext works |
| 03:19.51 | onixx | oh and while I'm here, anyone know of a good source for Grandstream... please let me know |
| 03:19.54 | Hydroxide | but I needed both the /n and the g |
| 03:20.03 | Hydroxide | onixx: I got mine from voipstore.pulver.com |
| 03:20.17 | *** join/#asterisk i-am (~AIntel@adsl-69-151-8-179.dsl.stlsmo.swbell.net) |
| 03:20.23 | nic_oz | pfn: sip debug doesn't show me any activity when I press the button, is there anything else I can try ? |
| 03:21.06 | iVoice | Anyone can help me about playing a stream at the same time in two channels? I'm writing a patch. Using ast_streamfile(chA, file, lang)/ast_waitstream(chA, "") and ast_streamfile(chB, file, lang)/ast_waitstream(chB, "") I have to wait stream on channel A to be finished before * play stream on ch b |
| 03:21.08 | Hydroxide | ok, thanks all .. bye |
| 03:21.11 | *** part/#asterisk Hydroxide (user@adsl-98-2.bway.net) |
| 03:23.14 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 03:24.46 | *** join/#asterisk agray (~agray@ip68-6-112-134.sb.sd.cox.net) |
| 03:25.48 | agray | I have a question from my telco about how many digits does my pbx expect to recieve. What is the right answer? |
| 03:26.11 | i-am | 69? |
| 03:26.31 | i-am | no, that's not right |
| 03:26.51 | agray | Nice... I did not assume do. |
| 03:26.55 | mikegrb | 70 |
| 03:27.39 | i-am | perhaps asking the telco what the usual response is would help |
| 03:27.57 | i-am | from what I've read, the asterisk software can work with a number of different switch types |
| 03:28.02 | agray | Ok. Just did not know if there is a standard or not. |
| 03:28.19 | Qwell | 7, 10, 11, 42...does it matter, really? |
| 03:28.43 | agray | Don't know, I am really new to the telco side of the house. |
| 03:28.56 | i-am | ask more questions |
| 03:29.06 | tzanger | what's the /n doing? |
| 03:29.07 | Qwell | ask them, "What does your pbx SEND?" |
| 03:29.08 | i-am | does the question mean how many digits are sent during an incoming call? |
| 03:29.08 | agray | the systems and networking makes sense to me... The telco seems like a strange place. |
| 03:29.38 | *** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com) |
| 03:30.01 | agray | Yes it is for incoming calls. |
| 03:30.17 | i-am | are you setting the pbx up now? |
| 03:30.24 | i-am | perhaps you can just try the standard setting |
| 03:30.35 | agray | No. Answering the form questions so I can order the new pri. |
| 03:30.43 | i-am | ah |
| 03:30.47 | Qwell | throw in a random number then |
| 03:30.52 | i-am | hehe |
| 03:30.56 | Qwell | "My PBX expects to receive 42 digits. No more, no less." |
| 03:31.03 | i-am | hehe |
| 03:31.04 | Qwell | give them a "wtf?" moment |
| 03:31.21 | agray | The telco company is pretty nice. I do not really want to screw with them. |
| 03:31.34 | i-am | asterisk works with 5ESS switches |
| 03:31.40 | i-am | and the DMS100s |
| 03:31.54 | agray | DMS100 seems to be pretty standard in our area. |
| 03:31.55 | i-am | call the phone company and ask for a little assistance |