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| 00:36.57 | p3nguin | wimpy: That old USB stick had the lock switch turned off, so I guess it was really FUBAR. I installed a new memory stick today, formatted it ext2 with no special options, and I mount it with defaults (which is rw,relatime,errors=continue). Do you have any recommendations for different options? |
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| 00:51.00 | mzb | noatime |
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| 00:59.32 | p3nguin | and nodiratime, as well? |
| 01:00.57 | p3nguin | This volume remains unmounted 99% of the time. When the cron job is ran, the volume is mounted, files and directories are written to the volume, then the volume is umounted again until another 24 hours. |
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| 01:15.49 | *** join/#asterisk devdvd (Jason@c-71-61-188-154.hsd1.wv.comcast.net) |
| 01:16.04 | devdvd | I'm looking for a good wisip phone. Suggestions? |
| 01:21.32 | p3nguin | Good, WiFi, SIP... Pick any two. |
| 01:24.07 | devdvd | ok let me rephrase, the best of the worst of the wifi sip phones :) |
| 01:24.22 | *** join/#asterisk rampage73 (~rampage73@isp.dctechonline.com) |
| 01:24.35 | rampage73 | can anyone help with trixbox here? |
| 01:25.51 | rampage73 | trying to make one trix box use the other for the gateway |
| 01:31.16 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 01:37.13 | p3nguin | devdvd: I've always favored Cisco. Would something like the 7921G with SCCP be out of the question? I know you asked for SIP, but there are other suitable channel technologies available. |
| 01:49.22 | devdvd | p3nguin, sccp isn't out of the question. How good is the sccp support in 1.6.2? |
| 01:49.52 | *** join/#asterisk shapr (~shapr@nat/digium/x-vtwytiuymxwevjxe) |
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| 01:51.27 | p3nguin | devdvd: I don't use the Asterisk-provided skinny channel driver because it wasn't very "complete" when I first checked it out in 1.4. I started using chan_sccp-b v2 and was very satisfied with my Cisco phone with an SCCP image. I later upgraded to the chan_sccp-b v3 branch and like it even more. |
| 01:52.17 | devdvd | very nice |
| 01:54.43 | devdvd | thanks p3nguin: i will check that out |
| 02:02.27 | *** part/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
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| 02:13.02 | p3nguin | Yay! I finally got my printer problem straightened out. |
| 02:15.01 | p3nguin | When using IPP via CUPS, the job would always fail with a cups backend error, and to get it to print I would have to dick around with resume and cancel. I determined that the print server would accept raw printing, so I changed cups to use socket:// rather than ipp:// and it works great again... finally. |
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| 02:19.27 | WIMPy | I haven't been able to print since I upgraded lpr to lpr-ng. And even CUPS knows both my printers I only get grabage out of it. |
| 02:28.08 | *** join/#asterisk bjhaid (~abejide@41.206.12.18.vgccl.net) |
| 02:33.35 | p3nguin | I lost my normal printing with a CUPS upgrade quite some time ago. I had to start using this stupid workaround. I'd make sure the printer said it was ready, then send the print job... once it got there, it showed a failure and said printer is not ready... then I would resume printer, which made the pages print and then go back to not ready. I would then cancel the job. |
| 02:33.57 | p3nguin | It was stupid, but at least I got my stuff to print eventually. |
| 02:34.34 | p3nguin | I finally had time to sit down and get it worked out so I can print like a normal person. |
| 02:35.13 | WIMPy | I get nor errors. Just pages of junk. But it's a high-level isue as I can cp any PS file to /dev/lp0 just fine. |
| 02:35.41 | p3nguin | Sounds like a driver problem. |
| 02:37.02 | WIMPy | Interestingly enough, bot printers support both PS an PCL, so in theory it should be hard to not get them working, but CUPS manages on both. NFI. |
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| 02:39.13 | WIMPy | Here I get away by just selecting 'generic PS' or 'generic LJ'. |
| 02:41.05 | p3nguin | direct connection or networked? |
| 02:41.40 | WIMPy | Parallel |
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| 02:42.41 | p3nguin | I'm sure mine would have been cake if I had the printer connected to my computer, but I have a network printer. |
| 02:43.42 | WIMPy | doesn't own such modern stuff. |
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| 02:56.35 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 02:57.15 | nny | hmm was told 1.8 includes adaptive_odbc but don't see it with installed digium package |
| 02:59.20 | *** join/#asterisk babbar (745ae072@gateway/web/freenode/ip.116.90.224.114) |
| 02:59.57 | babbar | hey I need the api for asteris 1.8 in c# |
| 03:00.09 | babbar | anu help?? |
| 03:00.27 | babbar | anyone!! |
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| 03:06.08 | nny | nm ignore me, package seperate |
| 03:06.42 | WIMPy | babbar: Can you be more vague? So far that does not sound like an Asterisk Question. |
| 03:07.09 | nny | and adaptive in odbc package, nice |
| 03:07.12 | babbar | yea |
| 03:07.37 | babbar | I downloaded the c# API Asterisk.NET |
| 03:07.51 | babbar | its not working for the version of asterisk 1.8 |
| 03:09.02 | babbar | Can you give me some reference where I can find documentations for asterisk event tracking via asterisk manager api |
| 03:10.04 | WIMPy | FOR the AMI stuff, I thin, voip-info.org might actually be a good place to start. |
| 03:12.12 | babbar | yea I downloaded the files for the AMI from that site |
| 03:12.18 | babbar | the documentation is good |
| 03:12.40 | babbar | it only supports asterisk < 1.8 |
| 03:12.44 | babbar | but the files are not working for asterisk 1.8 |
| 03:13.13 | babbar | I get an error "unknown asterisk version" |
| 03:13.47 | WIMPy | Contact the author or forget about it and use AMI yourself. Or just remove that check. |
| 03:14.33 | babbar | hmm |
| 03:15.10 | babbar | I need to code a client to track events and display a popup on an LINK event |
| 03:15.16 | babbar | any softwares that u know already exist |
| 03:15.32 | babbar | for these stuffs |
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| 03:16.28 | WIMPy | There's probab;y tons of thongs out there that do that or something similar. |
| 03:16.42 | WIMPy | damn. |
| 03:17.07 | babbar | hmm lemme check it out |
| 03:17.08 | babbar | thanks |
| 03:17.14 | WIMPy | But unless your typing is as bad as mine, just writing something might be quicker that searching. |
| 03:18.16 | WIMPy | He already knew which side of the keyboard faces up... |
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| 03:23.49 | *** join/#asterisk provolone (~root@216-164-49-134.c3-0.gth-ubr1.lnh-gth.md.cable.rcn.com) |
| 03:26.14 | provolone | pbx.c: Unable to register extension 's', priority 1 in 'from-bleh', already in use |
| 03:26.14 | provolone | How can I find where this is already used ? |
| 03:26.54 | WIMPy | Look at your dialplan? |
| 03:34.34 | iprouteth0 | cat your externsions.conf and grep the output |
| 03:35.21 | p3nguin | How about just grep your extensions.conf and save the cats for something more productive? |
| 03:35.37 | p3nguin | In other words, don't be a n00b. |
| 03:36.14 | shapr | I like cats. |
| 03:37.03 | p3nguin | The only good cat is a dead cat. |
| 03:41.30 | provolone | ive eliminated everything else from extensions.conf |
| 03:42.17 | p3nguin | "diaplan show s@from-bleh" should give you a clue. |
| 03:42.26 | provolone | its all of 5 lines and I am still seeing the same warning |
| 03:42.40 | p3nguin | Are you saving your changes and then running diaplan reload? |
| 03:43.27 | provolone | I am stoping the daemon and restarting it |
| 03:43.41 | p3nguin | Seems silly, but okay. |
| 03:44.06 | provolone | I am silly and green to telephony |
| 03:44.26 | p3nguin | Sorta like changing your clothes because you got some mud on your shoe. |
| 03:52.17 | provolone | I guess I dont know the difference between my foot-included rabbit ear pajamas |
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| 06:30.17 | _zoom_ | hello |
| 06:31.06 | _zoom_ | does asterisk in anyway support oneway password, for sip registration?? |
| 06:32.57 | p3nguin | What exactly do you mean and what are you trying to accomplish? |
| 06:34.09 | _zoom_ | each time client register will provide different secret, asterisk show follow that |
| 06:34.34 | _zoom_ | like SecureID |
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| 06:55.10 | jkroon | hi guys, when using call queues, it seems the only strategy affected by the agent penalty is the ringall? |
| 06:55.45 | jkroon | eg, leastrecent (at least) seems to ignore it, and the page at http://www.voip-info.org/wiki/view/Asterisk+call+queues implies it's only honored by ringall? |
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| 07:05.43 | *** join/#asterisk nicola_pav (~nicola@mail2.tikalnetworks.com) |
| 07:07.55 | nicola_pav | hello. i would like to detect call forward loops in asterisk |
| 07:08.09 | nicola_pav | any hints on what to start with? |
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| 07:16.29 | schmidts | good morning |
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| 07:21.13 | Dovid | morninn |
| 07:21.16 | shapr | hiay |
| 07:21.18 | shapr | hiya |
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| 07:42.57 | WIMPy | nicola_pav: What technology? |
| 07:44.22 | nicola_pav | WIMPy: sip |
| 07:45.55 | WIMPy | Hmm. The only easy thing would be to tweak a counter in to either number. the other thing would be to add a header. |
| 07:46.31 | nicola_pav | WIMPy: when i run database show cf in asterisk |
| 07:46.40 | nicola_pav | it returns all the CFs |
| 07:47.03 | nicola_pav | i thought of writing a script to detect multiple occurrences |
| 07:47.41 | WIMPy | So no SIP, but just local? |
| 07:48.09 | nicola_pav | sorry for the inconvenience |
| 07:48.24 | nicola_pav | i want to detect loops locally |
| 07:49.00 | nicola_pav | extensions leading to another extensions which in turn could lead back to the same first extension |
| 07:49.09 | nicola_pav | this creates a loop |
| 07:49.12 | wdoekes2 | nicola_pav: I use a list in which I store the destinations and check that if a destination is already in there |
| 07:49.12 | nicola_pav | right? |
| 07:49.17 | WIMPy | Ok, that's easy. Just set a variable when forwarding. Or increase a counter. |
| 07:49.34 | wdoekes2 | s/list/variable |
| 07:50.07 | nicola_pav | ok, i got the idea |
| 07:50.10 | nicola_pav | thanks |
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| 08:21.36 | beaver_rrr | hello; i have a problem with chan_skype outbound calls, i made a primitive dialplan where only calls for internal networks and one extension for skype_out exist, and cant figure out what it wands from me with "Auto fallthrough, channel 'SIP/111-00000002' status is 'CHANUNAVAIL'" |
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| 09:05.29 | allan8904 | hey, i was wondering if someone would be able to tell me how to map the number 0151 to 099168176 |
| 09:06.35 | shapr | You could match that input and then dial to that other number. |
| 09:06.47 | allan8904 | would tat be using the extensions.conf? |
| 09:06.54 | shapr | yes |
| 09:08.11 | allan8904 | hmm i'll give that a shot |
| 09:08.13 | allan8904 | thanks |
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| 10:10.59 | DelphiWorld | hey all |
| 10:11.11 | DelphiWorld | guys |
| 10:11.16 | DelphiWorld | iax2 is not longer working for me |
| 10:11.18 | DelphiWorld | any idea? |
| 10:12.26 | E-bola | restart your server? :) |
| 10:14.48 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 10:15.03 | beaver_rrr | exten => _749[95].,1,Dial(Skype/username@+7${EXTEN:1}) makes "Auto fallthrough, channel 'SIP/111-00000002' status is 'CHANUNAVAIL'". Whats wrong? |
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| 10:24.58 | Tozz_ | beaver_rrr: SIP peer 111 is not registered |
| 10:25.04 | *** join/#asterisk wonderworld (~ww@port-92-201-85-166.dynamic.qsc.de) |
| 10:25.05 | Tozz_ | or for some other reason unavailable |
| 10:25.17 | Tozz_ | oh sorry |
| 10:25.24 | Tozz_ | the SKype useris unavailable |
| 10:25.30 | flashdeluxe | Hi @ all! I am using asterisk 1.8 and everything is working fine, now i have a customer who wants to adjust several call forwards and i dont want to configure this by hand everytime he calls me, so i want to have a gui where he can configure it by himself, can sombody advice me a simple gui which is working with 1.8 ? |
| 10:26.01 | Tozz_ | all GUI's suck |
| 10:26.17 | WIMPy | And the GUI will have to fit your dialplan. |
| 10:26.42 | WIMPy | (or the other way round) |
| 10:27.10 | flashdeluxe | but it sucks to configure everything by hand for every customer whos calling you, too :( |
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| 10:27.34 | DelphiWorld | flashdeluxe: fusionpbx |
| 10:27.36 | DelphiWorld | flashdeluxe: blue.box |
| 10:27.44 | WIMPy | uses the usual feature codes. |
| 10:28.13 | DelphiWorld | shitttttttttttttttttttt |
| 10:28.13 | DelphiWorld | WIMPy: i'm crazy today |
| 10:28.27 | DelphiWorld | WIMPy: i answered it a freeswitch question :) |
| 10:28.30 | WIMPy | Well, if it's only today... :) |
| 10:29.04 | WIMPy | actually hates feature codes, but there isn't much you can do with Asterisk. |
| 10:29.05 | kaldemar | flashdeluxe: make an extension that toggles forwards how they want. :P |
| 10:29.18 | *** join/#asterisk AviMarcus (~avi@bzq-79-180-184-130.red.bezeqint.net) |
| 10:30.15 | flashdeluxe | kaldemar: maybe thats the way i have to do it... but for customers its easier to have a gui where they can adjust things |
| 10:30.49 | WIMPy | I started one, but never continued. |
| 10:31.18 | WIMPy | But that was for a lot more things. Maybe I should split it up and revive it. |
| 10:32.03 | AviMarcus | isn't there freepbx and other guis? |
| 10:32.04 | WIMPy | But I just read that the development branch of LCR was fixed, so I will concentrate on that one today. |
| 10:32.16 | AviMarcus | you didn't have working LCR? |
| 10:32.39 | WIMPy | "the development branch" |
| 10:32.52 | WIMPy | The asterisk_1_8 branch is working. |
| 10:32.57 | AviMarcus | ah k |
| 10:33.13 | WIMPy | Although I'm currently using something inbetween. |
| 10:33.34 | flashdeluxe | my problem is, that i don`t want to waste my dialplan, but if i use a gui i have to rewrite it and thats just annoying :/ |
| 10:34.50 | WIMPy | flashdeluxe: It shouldn't be too hard to du a simple web form yourself. |
| 10:36.29 | flashdeluxe | WIMPy: Yeah, but theres soo much to do and so less time :( but months ago i tested something called safi server, maybe i will use that, it didn`t look bad.. |
| 10:37.25 | WIMPy | Ja, the usual problem. |
| 10:38.12 | E-bola | I just setup an extension they can change |
| 10:38.28 | E-bola | or if its for a single person, i let them do it on their phones interface (we use snom's) |
| 10:38.40 | E-bola | Works fine, and is just as fast as a GUI |
| 10:39.15 | WIMPy | E-bola: You mean via the browser or by letting the phone do the forwarding? |
| 10:40.11 | *** part/#asterisk AviMarcus (~avi@bzq-79-180-184-130.red.bezeqint.net) |
| 10:40.23 | flashdeluxe | E-bloa i will do it by dialing a few digits, i guess, they are using grandstreams, shouldn`t be so hard for them to activate a CF by typing in digits |
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| 10:46.31 | E-bola | WIMPy: I have solutions where the users change the number asterisk is forwarding to by dialing an extension and entering the number |
| 10:47.08 | E-bola | We also have clients who change the forwarding number on their phones, and let the snom phone handle the forwarding, controlled either via the phone menu on the display or via the phone's browser interface |
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| 10:51.43 | DelphiWorld | yo Trixboxer! |
| 10:52.06 | Trixboxer | o/ |
| 10:53.27 | DelphiWorld | Trixboxer: sory but what this smil mean? o\ |
| 10:53.56 | beaver_rrr | high five. |
| 10:57.03 | flashdeluxe | if you change the o and the \, DelpiWorld, it wouldn`t be good if you are german :D |
| 10:58.46 | beaver_rrr | flashdeluxe: http://explosm.net/comics/2330/ ! |
| 10:59.16 | Trixboxer | DelphiWorld: o/ = hi :) |
| 11:00.13 | flashdeluxe | beaver_rrr: :D |
| 11:00.44 | beaver_rrr | look, urls are so correct! o/ o/ |
| 11:15.45 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-fiisuzmzrnvtlgrc) |
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| 12:06.06 | BlackBishop | any way I can reject a call ? |
| 12:06.14 | BlackBishop | hangup() should send the busy signal, right ? |
| 12:06.26 | schmidts | BlackBishop Busy send the busy signal ;) |
| 12:06.36 | schmidts | s/Busy/Busy() |
| 12:07.54 | BlackBishop | what does hangup send then ? |
| 12:12.20 | wdoekes2 | BlackBishop: hangup takes an optional argument |
| 12:14.10 | wdoekes2 | what it sends as default, I don't know. it may depend on certain state as well |
| 12:15.50 | wdoekes2 | see hangup_cause2sip (in case you're dealing with SIP) and the AST_CAUSE_ defines in causes.h |
| 12:15.51 | leifmadsen | but it's not busy by default :) |
| 12:16.22 | leifmadsen | there is Busy() and Congestion() typically for those things |
| 12:16.49 | wdoekes2 | congestion is not nice.. if that yields a 5xx, peers may think the server is down |
| 12:17.53 | beaver_rrr | http://pastebin.com/1m43bNpw -- whats wrong with this dialplan, that can preserve SFA from working? |
| 12:18.21 | BlackBishop | neah, I'm trying to reject users incomming calls through a datacard |
| 12:18.22 | coppice | if things are congested the server *is* functionally down |
| 12:19.16 | E-bola | datacard as in a wireless internet access? |
| 12:19.55 | BlackBishop | E-bola: datacard as in a huawei k3520 gsm voice modem accessible through chan_datacard |
| 12:20.06 | leifmadsen | beaver_rrr: uhhh.... so you want to call xxxxxxx@+7<whatever you dialed less the first digit> ? |
| 12:20.09 | E-bola | ohh serverside |
| 12:20.18 | leifmadsen | beaver_rrr: because that's essentially what you're dialplan says |
| 12:20.28 | beaver_rrr | yes, that is what i want |
| 12:20.38 | leifmadsen | beaver_rrr: but that seems wrong |
| 12:21.19 | leifmadsen | unless I guess you're inputing an IP address..... |
| 12:21.32 | leifmadsen | anyways sorry, someone else will have to look |
| 12:21.36 | beaver_rrr | http://pastebin.com/RLXx3xbg |
| 12:21.42 | beaver_rrr | there is what i got |
| 12:21.50 | leifmadsen | runs off to make significant amounts of coffee, eat breakfast, then try to get ready to work on a book |
| 12:22.32 | leifmadsen | beaver_rrr: ya that's not right.... you're calling xxxxxxxx@<peer called +74..... in skype.conf> I'm pretty sure |
| 12:22.47 | leifmadsen | I'm not 100% sure on the method Skype uses, but likely you want something more like |
| 12:22.57 | leifmadsen | Dial(Skype/some_skype_name) |
| 12:22.58 | *** join/#asterisk imox1234 (~imox1234@p4FC5C747.dip0.t-ipconnect.de) |
| 12:23.19 | leifmadsen | what you're trying to do looks terribly bizarre |
| 12:23.24 | beaver_rrr | leifmadsen: that is not just skype, that is skype_out |
| 12:23.28 | beaver_rrr | it can call PSTN |
| 12:23.34 | leifmadsen | yes I realize that |
| 12:23.52 | leifmadsen | it looks strange... perhaps that is how they have it setup |
| 12:23.59 | leifmadsen | but it doesn't look right :) |
| 12:24.01 | beaver_rrr | if i dial +7xxxxxxxxxx directrly from skype, it works |
| 12:24.33 | leifmadsen | I'd have expected it to be more like: Dial(Skype/my_account/${EXTEN:1} or something |
| 12:24.35 | beaver_rrr | and there is what digium's manual says: |
| 12:24.41 | beaver_rrr | "exten => ...,1,Dial(Skype/james_bond@+12564286000" |
| 12:24.45 | leifmadsen | odd |
| 12:24.49 | leifmadsen | ok then I guess that's the format |
| 12:25.01 | leifmadsen | I don't use skype out |
| 12:25.25 | leifmadsen | other end must be rejecting it for some reason |
| 12:25.32 | leifmadsen | does that account have credit to place calls? guess so... |
| 12:25.44 | beaver_rrr | it has, i can call from skype |
| 12:25.46 | leifmadsen | runs off to make significant amounts of coffee, eat breakfast, then try to get ready to work on a book <-- foe real |
| 12:25.53 | beaver_rrr | and also first call worked |
| 12:26.05 | beaver_rrr | looks like it didnt properly hanged up in asterisk |
| 12:32.58 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
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| 12:45.49 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 12:45.59 | OlafsenM | hey guys |
| 12:46.22 | OlafsenM | question: after a while Asterisk hogs up CPU |
| 12:46.39 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 12:46.41 | OlafsenM | 'core show threads' shows many '0x7f46188bd950 iax2_process_thread started at [ 1308] chan_iax2.c find_idle_thread()' |
| 12:46.46 | Sertys | hey, guys |
| 12:46.53 | Sertys | does anyone know what's the status on chan_skype? |
| 12:47.07 | Sertys | is it blocked by skype nowadays? |
| 12:47.55 | BlackBishop | ~2[2011-02-21 14:41:53] WARNING[12988]: app_dial.c:1328 wait_for_answer: Unable to write frame |
| 12:48.01 | BlackBishop | this doesn't look good |
| 12:48.26 | BlackBishop | phone -> incomming datacard .. enter number .. -> outgoing datacard -> final destination |
| 12:48.32 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:48.32 | BlackBishop | if the first phone closes |
| 12:48.40 | BlackBishop | I get a log of unable to write frame |
| 12:48.48 | BlackBishop | and the final destination still rings |
| 12:50.02 | BlackBishop | any ideas anybody ? |
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| 13:07.45 | BlackBishop | Sertys: I don't think it's good if http://www.chanskype.com/ reports 404 |
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| 13:36.12 | *** join/#asterisk topriddy (~Seamfix@41.58.47.161) |
| 13:36.32 | topriddy | hello people |
| 13:37.34 | topriddy | i have this lil problem i want to solve. need to automatically place calls to a list of numbers, play a pre-recorded message on picking d call, and then dial next number. i have about 200,00 numbers |
| 13:38.08 | topriddy | i understand at some point i may require connecting a phone modem with sim??? also do i have to get a PBX box too??? |
| 13:38.22 | topriddy | my question actually is what hardware would be required??? |
| 13:40.46 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 13:42.14 | topriddy | anybody care to help? |
| 13:44.43 | kaldemar | ~newbook |
| 13:44.43 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 13:45.14 | kaldemar | topriddy: many ways to do it, you don't necessarily need any telephony hardware, just an asterisk box if you use an ITSP. |
| 13:45.19 | topriddy | kaldemar: do tell me the hardwares i'll be needing, promise to look the book later |
| 13:46.12 | kaldemar | a PC with a NIC is the minimum. |
| 13:46.13 | *** part/#asterisk Akiraa (~Akira@82.76.239.176) |
| 13:47.07 | topriddy | kaldemar: why do i need NIC? |
| 13:47.38 | topriddy | am talking about GSM modem calls, dont want to rely on unreliable VOIP calls |
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| 14:08.28 | *** join/#asterisk commmmodo (~commmmodo@68.49.225.72) |
| 14:09.57 | commmmodo | hello everyone! |
| 14:11.20 | zamba | hi there! |
| 14:11.40 | commmmodo | good morning |
| 14:12.14 | commmmodo | i am trying to get some help in setting up an asterisk box |
| 14:13.49 | commmmodo | i made this forum post, but I didn't get a reply: http://forums.digium.com/viewtopic.php?f=13&t=77142&sid=c5013c034cdf1cb8078cf9c6992c4da0 |
| 14:17.09 | *** part/#asterisk fauxalliance (~fauxallia@142.162.197.28) |
| 14:17.13 | benngard | ~phones |
| 14:17.14 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
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| 14:26.40 | topriddy | i am looking at this for my solution: http://www.portech.com.tw/p3-product1.asp?Cid=6 |
| 14:26.47 | topriddy | any better suggestions? |
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| 14:31.56 | skrusty | afternoon |
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| 14:41.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:43.24 | pabelanger | topriddy: build your own? |
| 14:46.07 | topriddy | pabelanger: build what??? |
| 14:47.35 | pabelanger | Your own VoIP / GSM gateway? Not sure what you want to do |
| 14:48.04 | topriddy | <PROTECTED> |
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| 14:48.37 | topriddy | pabelanger: of course i'll be needing a form of hardware device for that.# |
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| 14:59.29 | beaver_rrr | anubody configured Skype For Asterisk here to work with landline numbers? |
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| 15:07.12 | *** join/#asterisk nathan7 (nathan@unaffiliated/nathan7) |
| 15:11.02 | nathan7 | Hi, a friend of mine just installed asterisknow |
| 15:11.21 | nathan7 | And he wants to use his modem for incoming and outgoing calls |
| 15:11.27 | nathan7 | How would one configure that? |
| 15:12.15 | Mhaddog | modems do not do that |
| 15:12.30 | Mhaddog | you need to get and analog card with fxo's ports.... |
| 15:12.43 | Mhaddog | there are several out there, digiums, sangoma, etc |
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| 15:15.43 | leifmadsen | nathan7: or an ATA (analog telephony adapter) that takes the signal from the phone line and turns it into SIP |
| 15:16.00 | leifmadsen | but no, you can't use your modem as an FXO adapter for Asterisk. |
| 15:16.16 | leifmadsen | (the cheaper option is to get a pay-as-you go SIP account or something) |
| 15:16.17 | nathan7 | mhm |
| 15:16.31 | leifmadsen | or pre-paid account |
| 15:16.43 | nathan7 | Another friend of mine said it'd probably work if it did TAPI or something |
| 15:16.48 | leifmadsen | nope |
| 15:16.56 | nathan7 | It's a voice modem, I have no idea where it came from |
| 15:17.05 | leifmadsen | it's not useful to you for Asterisk use |
| 15:17.18 | nathan7 | Mhm |
| 15:17.19 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 15:18.50 | nathan7 | niekie: damn you |
| 15:19.03 | nathan7 | =p |
| 15:19.33 | nathan7 | Hrm, so, what'd the cheapest adapter cost? |
| 15:19.43 | nestAr | ~$0 |
| 15:19.45 | nestAr | err |
| 15:19.48 | nestAr | ~$40 |
| 15:20.24 | nathan7 | That's a PCI card I assume? |
| 15:20.34 | nestAr | oh, i was talking ata |
| 15:20.36 | nestAr | external |
| 15:20.42 | nestAr | dunno about pci cards.. |
| 15:20.46 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:20.55 | nestAr | once upon a time i had a $10 lucent winmodem that "worked" |
| 15:20.57 | nathan7 | Mhm |
| 15:21.01 | nestAr | but it sucked pretty badly |
| 15:21.03 | nathan7 | I have a winmodem |
| 15:21.12 | nathan7 | Well, I sold it to my friend |
| 15:21.13 | nestAr | callid didn't work, call quaility was shady at best |
| 15:21.24 | nestAr | i think the chipset was overheating |
| 15:22.07 | nathan7 | niekie: ..but it's *POSSIBLE* :D |
| 15:23.06 | nathan7 | erm, nestAr* |
| 15:23.10 | nathan7 | niekie: oops |
| 15:23.43 | WIMPy | wonders why modem support was dripped, when other analog hardware is still supported. |
| 15:23.57 | leifmadsen | nathan7: it's possible with a very specific type of chipset on a modem, but support is not existent, and the amount of time you'll spend fighting it is not worth it |
| 15:24.08 | nathan7 | leifmadsen: aha |
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| 15:24.18 | leifmadsen | WIMPy: because the support costs outweigh the price you can justify selling the modem for |
| 15:25.03 | WIMPy | Hmm. In the time of modems it used to work quite well, i.e. 20-15 years ago or so. |
| 15:25.03 | leifmadsen | it just doesn't make any economic sense |
| 15:25.20 | leifmadsen | if it did, then people would do it |
| 15:25.40 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:25.47 | WIMPy | I did it and I know others who did it. |
| 15:25.55 | leifmadsen | I'm not debating this |
| 15:26.07 | WIMPy | And it certainly makes sense to use something you already got. |
| 15:26.30 | leifmadsen | I'm saying, it makes no sense for the people who would sell a modem and support that modem along with driver development |
| 15:26.42 | leifmadsen | viewpoint of the company, not viewpoint of the consumer |
| 15:26.58 | WIMPy | And I still used modem style interface until I changed to Asterisk 4 years ago. |
| 15:26.58 | leifmadsen | you applied my "it doesn't make any sense" to the wrong entity |
| 15:27.06 | *** join/#asterisk mach-bar (~mach-bar@xdsl-89-0-47-58.netcologne.de) |
| 15:27.31 | WIMPy | had the impression it was pretty standard these days. |
| 15:27.54 | leifmadsen | what? selling modems and doing driver development against them? You'd be severely wrong in my estimation. |
| 15:28.12 | WIMPy | AT+FCLASS=8 and you get an ADPCM stream. |
| 15:28.25 | leifmadsen | so how about them Leafs? |
| 15:28.39 | WIMPy | Leafs? |
| 15:28.44 | leifmadsen | Toronto Maples LEafsz |
| 15:28.47 | leifmadsen | Leafs* |
| 15:28.48 | Qwell | I dislike the Leafs, and anything that sounds similar. |
| 15:28.52 | leifmadsen | Qwell: burn |
| 15:30.02 | WIMPy | The whole I4L interface was designed to be compatible to modems so that you could recycle your software. |
| 15:30.26 | Qwell | I4L = ISDN |
| 15:30.58 | WIMPy | /dev/ttyI* to be precice |
| 15:32.15 | skrusty | would anyone here be interested in beta testing an online queue stats and wallboarding service? |
| 15:32.48 | leifmadsen | skrusty: potentially :) |
| 15:32.53 | skrusty | :) |
| 15:33.59 | nathan7 | A linksys PAP2 would work? |
| 15:33.59 | skrusty | it's in early development, and looking for input from testers, including stats, security and access methods etc. we have an alpha build at the moment, but looking for people who would have some heft stats to run and who would be interested in testing the service for us |
| 15:34.29 | nathan7 | ohwait |
| 15:34.30 | skrusty | only looking for 5 or so testers at the moment with one or more asterisk servers |
| 15:35.01 | skrusty | but if you'd be interested, pm me your email address and we can get you in :) |
| 15:36.14 | leifmadsen | skrusty: leif@leifmadsen.com :) (it's no secret) |
| 15:36.19 | nathan7 | So, what is a cheap ATA with FXO? |
| 15:36.26 | skrusty | cool :) |
| 15:36.39 | leifmadsen | nathan7: any of the linksys devices seem to work well |
| 15:36.51 | nathan7 | leifmadsen: Not all of them have FXO |
| 15:36.53 | drmessano-lt | SPA-3102 is the only one with FXO |
| 15:37.30 | drmessano-lt | It works, just needs a little tuning |
| 15:38.01 | leifmadsen | nathan7: then I suggest using any of the ones with FXO.... I thought that was kind of implied when I said any of the Linksys devices work well |
| 15:38.23 | commmmodo | can anyone shed some light on this forum topic? http://forums.digium.com/viewtopic.php?f=13&t=77142&sid=c5013c034cdf1cb8078cf9c6992c4da0 |
| 15:38.25 | nathan7 | mh-hm |
| 15:38.35 | commmmodo | i'm trying to set up a call forwarding & recording system |
| 15:38.49 | nathan7 | I recall having some FXO-enabled ATA once |
| 15:39.04 | nathan7 | but that was free because my mom worked at Vonos or something |
| 15:39.14 | nathan7 | Vonos doesn't seem to exist anymore |
| 15:39.35 | Qwell | vonage? |
| 15:39.41 | nathan7 | No, not vonage. Vonos. |
| 15:39.43 | WIMPy | commmmodo: That's surely possible. |
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| 15:40.54 | commmmodo | WIMPy thanks! i am still reading up about asterisk and learning... how would i go about something like this? I asked some followup questions on the forum but it's been dead since then... :( |
| 15:41.03 | WIMPy | commmmodo: But in most countries there will be some legal stuff needed sorting, so you might better start asking a lawyer before getting to the technical part. |
| 15:41.22 | commmmodo | i have it taken care of, i just need the technical part |
| 15:41.26 | *** part/#asterisk beaver_rrr (~beaver@80.90.116.248) |
| 15:42.03 | WIMPy | commmmodo: Ok, first you need to find out how you get your calls in and out. PSTN or VOIP? |
| 15:42.34 | WIMPy | That requires either hardware or a service provider, respectively. |
| 15:42.43 | commmmodo | WIMPy: what would you recommend for this situation? |
| 15:43.20 | commmmodo | WIMPy: I would prefer VOIP |
| 15:43.31 | WIMPy | I don;t know what kind of service you can get where you are, so it's hard to say. |
| 15:43.46 | WIMPy | Or at what price. |
| 15:43.53 | WIMPy | ~ITSP |
| 15:43.53 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 15:44.01 | commmmodo | WIMPy: what are some of the service providers I can get VOIP through? |
| 15:44.04 | commmmodo | gotcha |
| 15:45.10 | commmmodo | ~itsplist-us |
| 15:45.10 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 15:46.16 | commmmodo | WIMPy: wow, very helpful, let me read over this a second |
| 15:59.41 | jaytee | I use Flowroute and I've been very satisfied with their service and pricing |
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| 16:35.38 | commmmodo | jaytee: are there monthly charges or is flowroute just charged by the minute? |
| 16:38.27 | *** join/#asterisk Godfather_ (~godfather@90.162.102.229) |
| 16:38.47 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:44.18 | p3nguin | commmmodo: It depends if you have recurring services or not. If you have a DID or a virtual PRI, you'll pay monthly fees. If you only have termination, you'll pay for usage only. |
| 16:56.48 | *** join/#asterisk mateu (~mateu@suryahunter.com) |
| 17:00.10 | commmmodo | p3nguin: cool thanks |
| 17:00.36 | *** join/#asterisk nny1 (~Scott_2@cpe-174-107-201-103.sc.res.rr.com) |
| 17:01.13 | nny1 | having an issue where when MeetMe is called, the system just exits nonzero, and I don't see the application being attempted or used |
| 17:01.28 | p3nguin | commmmodo: VoIP.ms is the same way. |
| 17:03.25 | commmmodo | p3nguin: which service do you use personally? |
| 17:04.02 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 17:05.01 | p3nguin | commmmodo: I primarily use VoIP.ms, but I dabble with several ITSPs. |
| 17:05.31 | commmmodo | p3nguin: how is VoIP.ms' uptime? is it reliable? |
| 17:05.45 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:06.59 | p3nguin | commmmodo: I have to assume their uptime is fine because I rarely have any problems making and receiving calls. When I do have problems, I figure it's likely a connection problem between my system and theirs. |
| 17:08.13 | commmmodo | p3nguin: do you use them for origination or termination or both? |
| 17:10.02 | nny1 | exten => 8101,1,MeetMe(8101) |
| 17:10.11 | nny1 | that's my test, isn't that right? |
| 17:10.50 | *** join/#asterisk mach-barer (~mach-bar@xdsl-89-0-22-218.netcologne.de) |
| 17:11.27 | mach-barer | has someone a sangoma b700 installation guide for dahdi? I can only find installation guides using zaptel... please query |
| 17:12.01 | p3nguin | commmmodo: both |
| 17:13.10 | p3nguin | nny1: If 8101 is not configured in meetme.conf, no... you need to use the correct option to create the conference dynamically. |
| 17:15.15 | p3nguin | I might use something like exten => 8101,1,MeetMe(${EXTEN},dMops); |
| 17:16.55 | nny1 | p3nguin: it is, with no options in meetme.conf |
| 17:17.09 | nny1 | p3nguin: any reason why no error *at all* is provided, and the system just exits? |
| 17:17.41 | nny1 | p3nguin: from meetme.conf "conf => 8101," |
| 17:18.19 | nny1 | p3nguin: troubleshooting something that worked previous here, my test is only to remove the current method meetme is called |
| 17:18.47 | nny1 | p3nguin: but the "exits non zero" and no feedback when it hits that part of the dialplan is infuriating |
| 17:19.55 | p3nguin | Exits non-zero isn't necessarily indicative of an error. |
| 17:20.18 | leifmadsen | agreed |
| 17:20.20 | nny1 | p3nguin: looks like the application is straight upnot working |
| 17:20.24 | nny1 | well |
| 17:20.38 | leifmadsen | where is the console output and full configuration details? please pastebin |
| 17:20.56 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 17:20.56 | leifmadsen | version you're using? you've confirmed the application is loaded? |
| 17:21.04 | p3nguin | core set verbose 4, call your meetme, share the details. |
| 17:21.11 | leifmadsen | more information provides better answers |
| 17:27.08 | nny1 | http://pastebin.com/ndn9tYMR |
| 17:27.14 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 17:27.17 | nny1 | asterisk 1.8 |
| 17:28.06 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:28.24 | nny1 | http://pastebin.com/VYtbkUSY |
| 17:28.36 | nny1 | with asterisk showing the relative context |
| 17:29.32 | nny1 | forcing sip-peer into ext-meetme |
| 17:29.38 | nny1 | via context in sip.conf |
| 17:29.41 | nny1 | to cimplify testing |
| 17:29.43 | nny1 | simplify* |
| 17:29.46 | nny1 | leifmadsen: ^^^ |
| 17:30.03 | p3nguin | ("Next line is Meetme") <--- this is not a valid application or function. |
| 17:30.25 | nny1 | p3nguin: oh yeah, just added that for testing, one sec, missed NoOp |
| 17:32.11 | nny1 | p3nguin: leifmadsen http://pastebin.com/nysZBPx0 |
| 17:32.36 | nny1 | like i mentioned before non-zero, nothing, dead, killed, dusted, broken, etc |
| 17:32.59 | nny1 | hits exten => 8101,n,MeetMe(8101) and shits the bed |
| 17:33.16 | nny1 | sorry |
| 17:33.20 | nny1 | asterisk version helps |
| 17:33.31 | nny1 | Asterisk 1.8.2.2 built by root @ localhost.localdomain on a x86_64 running Linux on 2011-01-20 21:32:01 UTC |
| 17:33.35 | p3nguin | Did you confirm that app_meetme.so was loaded like leifmadsen suggested? |
| 17:34.08 | nny1 | p3nguin: sure isn't |
| 17:34.12 | nny1 | p3nguin: why would that happen |
| 17:34.27 | nny1 | there isn't any noload line for it in modules.conf |
| 17:34.34 | p3nguin | Maybe you didn't have a valid conf when you started asterisk, or maybe you don't use autoload. |
| 17:34.37 | drmessano-lt | He stated in #freepbx he had run some updates.. I suspect he updated the kernel and rebooted, broke DAHDI |
| 17:34.45 | nny1 | drmessano no i didn't |
| 17:34.57 | drmessano-lt | Well, you never answered any of us |
| 17:35.39 | p3nguin | Try loading the module manually and see what happens. |
| 17:35.53 | p3nguin | If it loads, dahdi is probably okay. |
| 17:36.02 | p3nguin | Well, I take that back... |
| 17:36.04 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
| 17:36.27 | p3nguin | I've been able to load MeetMe and even run the MeetMe() without dahdi working. |
| 17:36.31 | nny1 | doesn't look like dahdi is loaded, it's installed from packages |
| 17:36.41 | p3nguin | It just fails with error if dahdi isn't working. |
| 17:36.58 | nny1 | yeah can't run dahdi show in cli |
| 17:37.16 | p3nguin | Take care of that, then try app_meetme again. |
| 17:37.21 | drmessano-lt | chkconfig dahdi on && service dahdi start |
| 17:38.05 | Kobaz | what's a little hardware box that can behave as a windows vpn client |
| 17:38.08 | drmessano-lt | [11:10] <RypPn> does dahdi_test work? Only time I ever see confs hangup is when dahdi isn't loaded |
| 17:38.08 | Kobaz | (pptp) |
| 17:38.12 | nny1 | yeah started it, amporatl doesn't start it.. meh |
| 17:38.25 | nny1 | but no error etc is annoying |
| 17:38.35 | nny1 | have to restart asterisk |
| 17:38.45 | nny1 | waiting for an open moment |
| 17:38.54 | nny1 | (knows about restart when) |
| 17:40.23 | p3nguin | I like restart gracefully. |
| 17:40.35 | p3nguin | That keeps people from making new calls. |
| 17:40.45 | p3nguin | restart when convenient allows new calls. |
| 17:40.47 | nny1 | so exits non-zero is normal when dahdi dummy isn't running? |
| 17:40.56 | nny1 | yeah have to alert them to it first |
| 17:44.04 | nestAr | <PROTECTED> |
| 17:44.19 | nestAr | I see calls coming in, but they get dropped almost immediately |
| 17:44.24 | nestAr | any hints? |
| 17:44.26 | nestAr | PRI |
| 17:44.59 | WIMPy | {0x06, "Channel unacceptable"}, |
| 17:45.17 | nestAr | yeah,. i'm just wondering if it's something in my config |
| 17:45.21 | nestAr | or the phone company |
| 17:45.22 | WIMPy | Is it a partial PRI or something? |
| 17:45.27 | nestAr | yeah.. |
| 17:45.32 | nestAr | 5 b channel |
| 17:46.05 | WIMPy | Then you tried to access a channel you didn't pay for. |
| 17:46.27 | nestAr | k |
| 17:47.05 | WIMPy | Either you configured more than those 5 channels or you got a 6th call. |
| 17:48.25 | WIMPy | Hmm. If you have onl those 5 channels configured, I guess the 6th call should give some other error, like 0x22. |
| 17:48.56 | *** join/#asterisk javier_cintron (~javier_ci@200-56-202-1-cable.cybercable.net.mx) |
| 17:50.03 | nny1 | p3nguin: started dahdi, nothing |
| 17:50.11 | nny1 | p3nguin: dahdi loaded dummy channel driver |
| 17:50.20 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 17:50.55 | nestAr | what's cause 100? |
| 17:51.31 | nny1 | p3nguin: dahdi show doesn't work still |
| 17:51.36 | WIMPy | {0x64, "Invalid information element contents"}, |
| 17:53.22 | nestAr | phone company is blaming my pbx, which im ok with. im not the smartest guy, but hes not giving me anything to go on |
| 17:54.17 | WIMPy | 'pri set debug 2 span 1' and pastebin the output. |
| 17:54.27 | WIMPy | (or read it yourself) |
| 17:54.58 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:449e:b571:97e7:d0cc) |
| 17:55.38 | nny1 | how do I diagnose dahdi not loading even after the init.d script is run and all reports "ok"??? |
| 17:55.50 | javier_cintron | Hi there. I am installing a asterisk pbx, what version is the best in my case? I read in a blog that new versions like 1.8 are unstable and that is better to use 1.6 for a production server. what do you think? |
| 17:56.27 | WIMPy | has upgraded all instances to 1.8. |
| 17:56.55 | nestAr | http://pastebin.com/KpPxkYc0 |
| 18:00.20 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 18:01.56 | nestAr | Mostly greek to me.. Maybe you guys have an idea. :D |
| 18:04.12 | commmmodo | WIMPy: i read into some of those ITSP's. what will i need to do after I sign up with one of them? |
| 18:04.12 | commmmodo | WIMPy: is setting up the call forwarding and recording just a matter of installing asterisk on a linux box and creating the routing for those numbers? |
| 18:04.22 | WIMPy | nestAr: The original SETUP message is missing. That's bad. But I can see Asterisk trying to accept the call on cahnnel 3 which does not seem to be a valid one. |
| 18:05.15 | WIMPy | commmmodo: Configure them in sip.conf. |
| 18:05.44 | WIMPy | Then you configure your functionality in extensions.conf and that's all for the main task. |
| 18:07.54 | nestAr | WIMPy: channel 3 doesn't appear to be setup on the phone company side? i have channels 1-5 enabled in my conf |
| 18:08.32 | WIMPy | That's what it looks like. But the missing forst message should give you a hint. |
| 18:09.48 | WIMPy | And there seems to be a timing issue as well, but that's not really important. |
| 18:09.55 | nestAr | is there anyway to log this debug to a file, to make sure i'm getting it all? maybe lost in the scroll back? |
| 18:10.29 | WIMPy | 'pri set debug file /tmp/debug' or something like that. |
| 18:10.32 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 18:13.16 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 18:13.21 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 18:13.58 | nestAr | WIMPy: http://pastebin.com/5YHP6YNn |
| 18:14.29 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 18:15.04 | drudge` | any recommendations on a sip trunk wholeseller? |
| 18:15.13 | Mhaddog | ?iax |
| 18:15.22 | Mhaddog | ?iax trunk |
| 18:15.26 | Mhaddog | ?? |
| 18:15.28 | Mhaddog | ? |
| 18:15.30 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 18:15.54 | citywok | ~itsp |
| 18:15.54 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:16.27 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:16.49 | drudge` | any recommendations tho.... |
| 18:16.53 | WIMPy | nestAr: Your number looks interesting. |
| 18:17.23 | drmessano-lt | drudge`, flowroute |
| 18:17.34 | citywok | ~itsplist-us |
| 18:17.34 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:17.46 | WIMPy | nestAr: So the call comes in on cannel 1 but Asterisk tries to accept it on cannel 3. I wonder why it would do so if you configured channels 1-5. |
| 18:17.49 | drudge` | we currently use Dash Carrier |
| 18:18.10 | drudge` | seeking better alternatives |
| 18:18.15 | citywok | awesome |
| 18:18.30 | drudge` | i dunno how awesome dash is... |
| 18:18.43 | ssureshot | WIMPy: just an fyi,,, there was a loopback in place on my router on friday.. time waisted on my end.. as all looked well untill I looked at the router |
| 18:18.49 | javier_cintron | does a atom duo core with 2 GB ram is enough for a pbx with 4 pstn lines and 10 internal normal(no IP) lines? We need voice mail and voice menu too |
| 18:18.49 | drudge` | they went compeltely down the start of the new year |
| 18:18.56 | citywok | awesome |
| 18:19.06 | citywok | javier_cintron: that's overkill :) |
| 18:19.20 | ssureshot | what OS does AsteriskNOW run on? |
| 18:19.22 | drudge` | they blamed it on uploading the wrong config to their SBC and their redundant-back-up SBCs |
| 18:19.27 | citywok | ssureshot: centos something |
| 18:19.29 | WIMPy | ssureshot: Yes, that sure looked like a loop. |
| 18:20.01 | ssureshot | I guess red means bad lol |
| 18:20.12 | WIMPy | nestAr: Err. There's something about the channel types going on, I think. |
| 18:20.28 | drudge` | drmessano, do you currently or have you used them in the past? |
| 18:20.40 | ssureshot | I assumed that the telephony co could have told me that but in all actuallity the only tested to the dmark and the problem was past that |
| 18:21.01 | ssureshot | citywok: thanks I might d/l that and take a look then |
| 18:21.23 | nestAr | maybe i configured something wrong....... |
| 18:21.29 | nestAr | i'm looking over it again |
| 18:21.40 | nestAr | not my first time with a pri, but it's been a while. |
| 18:22.00 | citywok | nestAr: on turnups i generally assume that it won't work until after a couple support calls with the telco |
| 18:22.22 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 18:22.35 | nestAr | citywok: right, i'm not panicing yet.. just kinda lightly flustered |
| 18:22.38 | citywok | nestAr: inevitably somebody left a loopback turned on somewhere. or once we had 5 turned on, that took the telco a day to figure out between 3 different international carriers speaking... not english. |
| 18:23.07 | WIMPy | nestAr: What version of libpri are you using? Maybe there's somethign going on. |
| 18:26.59 | nestAr | libpri-1.4.11.5 |
| 18:27.19 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 18:27.53 | Mhaddog | nestAr: what r u sugint o connect to the t1 ? |
| 18:27.59 | Mhaddog | using sorry |
| 18:28.32 | nestAr | digium TE122 |
| 18:29.26 | Mhaddog | u r in usa? or europe? |
| 18:29.29 | nestAr | US |
| 18:29.31 | nestAr | Texas |
| 18:29.54 | nestAr | cBeyond is the CLEC |
| 18:29.54 | Mhaddog | did u setup it correctly? |
| 18:30.04 | Mhaddog | meaning, ez8f or something and I forgot the ohter parameter... |
| 18:30.07 | WIMPy | nestAr: The cahnnel identification does look wrong, but I'm not sure yet, what it's trying to do. |
| 18:30.12 | Mhaddog | prob u r getting a PRI norrect? |
| 18:30.21 | Mhaddog | ok it is a single span card |
| 18:30.29 | Mhaddog | r u using all inbound? mixed? |
| 18:30.41 | Mhaddog | do you have the echo cancellation module on it? |
| 18:30.58 | nestAr | no, no echo cancel |
| 18:31.01 | Mhaddog | ok |
| 18:31.12 | Mhaddog | do you have any other card in your system? |
| 18:31.26 | nestAr | no |
| 18:31.28 | Mhaddog | any tdm? or sangomas? T1 or analog ones? |
| 18:31.29 | Mhaddog | ok |
| 18:31.32 | Mhaddog | so only this card |
| 18:31.33 | nestAr | that's the only pci card in the whole box |
| 18:31.42 | Mhaddog | that should be simply |
| 18:31.53 | Mhaddog | have you checked yout chan_dahdi.conf and system.conf files? |
| 18:31.53 | WIMPy | Mhaddog: The signalling link is working. Asterisk is just sending funny cahnnel identifications. |
| 18:32.16 | Mhaddog | have you checked that chan 24 is setup as the d channel? |
| 18:32.17 | nestAr | Mhaddog: yea, everything seems to be right, but i'd be happy to paste bin those as well. |
| 18:32.21 | Mhaddog | ok |
| 18:32.30 | nestAr | Mhaddog: yeah, that's what they're telling me. |
| 18:32.33 | Mhaddog | did u use the freepbx dahdi interface editor? |
| 18:32.46 | nestAr | no, no freepbx installed |
| 18:33.12 | Mhaddog | sorry I though i was int eh freepbx channel |
| 18:33.12 | Mhaddog | ok |
| 18:33.18 | Mhaddog | did u hand edited the install? |
| 18:33.35 | Mhaddog | pastebin the system.conf and chan_dahdi.conf please |
| 18:33.43 | Mhaddog | let me find mine to compare... |
| 18:34.08 | WIMPy | nestAr: I can't make any sense of it. It somehow looks like it doesn't fit together. I think you should try to upgrage libpri and retry. |
| 18:36.41 | nestAr | am i not on the latest already? |
| 18:37.17 | WIMPy | Or paste your dahdi configs as Mhaddog suggested. |
| 18:37.36 | Mhaddog | I'm just guessing if I do not see configs... |
| 18:38.01 | Mhaddog | I have not seen either a pastebin from cli with a fail call |
| 18:38.02 | nestAr | http://pastebin.com/u4Zy41m5 |
| 18:38.08 | nestAr | there's the dahdi |
| 18:39.09 | Mhaddog | why you have bchan 1-5? |
| 18:39.18 | nestAr | those are my lines |
| 18:39.22 | nestAr | i only have 5 lines |
| 18:39.50 | Mhaddog | ok |
| 18:40.20 | Mhaddog | I see different groups associated to the same trunk |
| 18:41.02 | Mhaddog | and i think i saw different spans too... |
| 18:41.05 | WIMPy | Try to comment out trunkgroup and spanmap. |
| 18:41.16 | Mhaddog | you said you only have one card.... |
| 18:41.25 | Mhaddog | I think ur configs are bad |
| 18:41.51 | nestAr | WIMPy: they're saying something about NFAS |
| 18:41.56 | nestAr | does that mean anything to anyone. |
| 18:42.08 | WIMPy | Erm, am I missing something or are there no channels in chan_dahdi.conf? |
| 18:42.27 | WIMPy | nestAr: Yes, but _they_ don't seem to use it. |
| 18:42.38 | Mhaddog | I dont know... he pasted all in one... |
| 18:42.48 | Mhaddog | I gave it a quick read let me go again |
| 18:43.05 | WIMPy | And it doesn't make any sense with only one physical interface. |
| 18:43.13 | nestAr | WIMPy: it's set to include dahdi-channels.conf |
| 18:43.28 | Mhaddog | I honeslty do not know why you need dahdi-channels... |
| 18:43.32 | WIMPy | ok |
| 18:43.43 | Mhaddog | and I have use a couple T1 with less than 5 channels |
| 18:43.46 | WIMPy | neither |
| 18:43.46 | Mhaddog | so I will say |
| 18:43.50 | Mhaddog | redo the whole thing |
| 18:44.00 | WIMPy | Less than 5 cahnnels? |
| 18:44.48 | WIMPy | You gouys should really find out what BRIs are. Using a PRI for so little must be quite expensive. |
| 18:45.08 | Mhaddog | sorry i meant more than 5 |
| 18:45.13 | Mhaddog | but not 23 at all |
| 18:45.32 | Mhaddog | yea a PRI pays of around 14 to 16 chan |
| 18:45.34 | Mhaddog | anyways |
| 18:45.40 | Mhaddog | i think your configs are forked... |
| 18:46.37 | Mhaddog | get to chan_dahdi and make it to look like |
| 18:46.44 | Mhaddog | let me see if i can text edit for you |
| 18:46.59 | Mhaddog | actually did you got this card new? |
| 18:47.30 | javier_cintron | Here in my work we want to change our very very old panasonic PBX(12 years old) with a new linux pbx. I wrote down all the functions this panasonic PBX is giving us to see what are simirlar functions on asterisk. The first one is call on hold: in our phones we simply push flash button and when we want to take this call again, we push flash again. ¿how is this function used in asterisk? |
| 18:48.48 | pabelanger | javier_cintron: you'll likely have to replace your current phones. Unless you plan to use the Panasonic PBX as a channel bank. |
| 18:48.50 | fenrus | placing a call on hold with a button ? |
| 18:49.40 | pabelanger | javier_cintron: but yes, Asterisk will likely do everything you need and more |
| 18:49.52 | Mhaddog | give me a sec nestAr |
| 18:50.09 | nestAr | trunkgroup and spanmap seemed to fix it. |
| 18:50.14 | nestAr | i'm making and taking calls. |
| 18:50.55 | javier_cintron | pabelanger: We want to replace this old panasonic PBX because is giving us a lot of problems. |
| 18:51.20 | javier_cintron | fenrus: Yes, we use flash button |
| 18:51.24 | pabelanger | javier_cintron: Okay, and your phones? |
| 18:51.40 | javier_cintron | fenrus: At least on our phones |
| 18:52.06 | fenrus | with modern sip phones you can have the same functionality |
| 18:52.38 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 18:52.53 | citywok | has anybody connects * with Exchange Unified Messaging? |
| 18:52.59 | citywok | s/connects/connected/ |
| 18:53.04 | javier_cintron | pabelanger: We plane to buy 2 sip phones but not to replace the others |
| 18:53.15 | Mhaddog | nestAr: you can check this http://pastebin.com/mtrwigWu if you want |
| 18:53.23 | Mhaddog | I just did it on the run, but I think should work |
| 18:54.04 | javier_cintron | pabelanger: but I was researching is our old phones work wirh asterisk or not |
| 18:54.25 | javier_cintron | with all function we need |
| 18:55.05 | pabelanger | javier_cintron: How do you plan to interface your previous phones to asterisk? You need to see if you even can. Most Panasonic phones are proprietary and will only with with Panasonic PBXs. If that is the case, you _could_ use the original PBX as a channel bank, otherwise you need to replace all your phones |
| 18:55.32 | pabelanger | I suspect you will need new phones |
| 18:55.51 | fenrus | its probably easier to not mix in the panasonic as a channel bank |
| 18:56.00 | fenrus | perhaps not cheaper |
| 18:56.47 | pabelanger | fenrus: That's the choice he needs to make. Save $$ and use previous phones, however it will not be as easy as new phones. But more $$$ |
| 18:57.02 | fenrus | pabelanger, yes of course.. |
| 18:57.16 | pabelanger | fenrus: http://en.wikipedia.org/wiki/Project_triangle :) |
| 18:57.21 | fenrus | pabelanger, =)) |
| 18:57.52 | javier_cintron | We are planning to use a Cisco SAP8800 with 4 pstn lines to interface with our old phones, what do you think? |
| 18:58.06 | javier_cintron | SOORY |
| 18:58.09 | javier_cintron | soory |
| 18:58.11 | javier_cintron | sorry |
| 18:58.15 | javier_cintron | my mistake |
| 18:58.31 | nny1 | so anyone have an official reason why asteriskcore, dahdi-tools and dahdi-linux don't include the actual channel driver for asterisk? |
| 18:58.52 | nny1 | i know they can be exclusive, but seems, superfluous |
| 18:59.14 | nny1 | (speaking of digium centos package repo, not sure about others) |
| 18:59.24 | pabelanger | javier_cintron: Why? What is the benefit of using the Cisco hardware? Why keep your original phones? Cost? |
| 18:59.28 | citywok | nny1: i just compiled & installed dahdi, and then asterisk, and meetme works so it looks native to me |
| 18:59.29 | nny1 | i was gonna submit some kind of bug on it, but realized it may be working as intended |
| 18:59.41 | nny1 | citywok: yeah this is from package, this install required it sadly |
| 18:59.54 | javier_cintron | We are planning to use a linksys SPA8000 to interface with our old phones, and a cisco sps8800 to interface with ouir 4 pstn lines. |
| 19:00.00 | javier_cintron | Yes cost |
| 19:00.10 | nny1 | citywok: usually afaik astierks compile inserts dahdi channel driver if dahdi is already compiled, installed |
| 19:00.21 | nny1 | but with packages it's standalone |
| 19:00.35 | citywok | oh, gotcha. don't use the packages :P |
| 19:00.36 | javier_cintron | Every sip phone here in Guadalajara, Mexico costs about 100 dollars, and we have 10 phones |
| 19:00.51 | javier_cintron | 1000 dollars |
| 19:01.02 | javier_cintron | spa8000 cost 250 dollars |
| 19:01.04 | citywok | javier_cintron: $100/phone is cheap |
| 19:01.16 | fenrus | thats some cheap phones |
| 19:01.41 | javier_cintron | with spa8000 we can interface 8 phones |
| 19:01.46 | javier_cintron | it is cheaper |
| 19:02.04 | ssureshot | provided the same features are enabled .. if I compile from src and copy the config files from version 1.2 to 1.8 should it work ? or are there many differences between the versions? |
| 19:02.16 | citywok | javier_cintron: lucky you, our old phones were inter-tel devices that were proprietary so an SPA couldn't have worked. we have 700 of them. |
| 19:02.28 | pabelanger | javier_cintron: I'd strongly consider the route of SIP phones. The more adapters you add into your VoIP network, the greater the difficulty to manage and troubleshoot is. Specifically if you don't have previous VoIP experience. |
| 19:02.32 | citywok | ssureshot: 1.2 to 1.8 there are a lot of changes |
| 19:03.00 | pabelanger | ssureshot: read the CHANGES and UPGRADE.txt in the asterisk-1.8 source directory |
| 19:03.07 | javier_cintron | citywork: my god :) |
| 19:03.18 | citywok | we use a handful of pap2's for old conference phones, etc. one of them is so sensitive if somebody hits a DTMF digit on a cellphone in the conference room, asterisk will hear it and meetme freaks out. |
| 19:03.19 | pabelanger | and UPGRADE-1.6.txt and UPGRADE-1.4.txt |
| 19:03.20 | ssureshot | will do,, thanks |
| 19:05.41 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 19:05.59 | citywok | ssureshot: i'd recommend doing a test upgrade not on your production box :) |
| 19:07.06 | ssureshot | citywok: no way lol.. all or nuffin!!! but no really I'm upgrading with new hardware and doing the research / testing.. |
| 19:07.16 | citywok | :P |
| 19:07.21 | citywok | just checking |
| 19:07.32 | ssureshot | hehe |
| 19:08.18 | javier_cintron | pabelanger: Certanintly you are right I dont have any experience with asterisk. I just conviced my boss to change this old pbx and he asked my for a quote. I am going to follow your advice and quote them |
| 19:08.35 | javier_cintron | pabelanger: too |
| 19:08.57 | citywok | javier_cintron: in that case definitely provide the SIP option and tell him it'll be much easier :) |
| 19:09.13 | citywok | a week of headaches is worth mroe than $750 |
| 19:09.16 | pabelanger | javier_cintron: Another option is to use something like switchvox, if you don't want to manage your own Asterisk server. |
| 19:11.22 | *** join/#asterisk dhorner_mb (~dhorner_m@pool-173-50-198-142.aubnin.fios.verizon.net) |
| 19:11.24 | nestAr | WIMPy: THANK YOU |
| 19:11.29 | nestAr | Mhaddog: THANK YOU |
| 19:11.37 | nestAr | to anyone else i forgot, THANKS. |
| 19:11.43 | Mhaddog | ur welcome |
| 19:11.46 | *** part/#asterisk dhorner_mb (~dhorner_m@pool-173-50-198-142.aubnin.fios.verizon.net) |
| 19:12.09 | nestAr | Once i got calls coming in, things got a little busy, so i got side tracked, but your help is much appreciated. |
| 19:12.09 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 19:12.56 | javier_cintron | pabelanger: I would like to install it by myself. At least for me is exciting to know all this asterisk stuff. :) |
| 19:13.16 | pabelanger | javier_cintron: +1 |
| 19:13.19 | pabelanger | welcome |
| 19:16.29 | *** part/#asterisk nny1 (~Scott_2@cpe-174-107-201-103.sc.res.rr.com) |
| 19:18.23 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 19:21.15 | javier_cintron | Ok thank you very much for your kind help. I am going to quote 10 ip Phones. To interface with our 4 pstn lines, we were planning to use a cisco spa8800, what do you think about this, is there a better solution? |
| 19:21.51 | Tozz_ | 8800? |
| 19:22.06 | javier_cintron | Yes |
| 19:22.12 | Qwell | javier_cintron: You could also use something like a Digium TDM410 (or TDM2400 if the 4 will ever expand) with 4 FXO ports. |
| 19:22.12 | javier_cintron | Cisco SPA8800 |
| 19:22.21 | Tozz_ | ah the gateway |
| 19:22.24 | Tozz_ | i thought that was a phone ;) |
| 19:22.40 | javier_cintron | :) |
| 19:22.52 | Tozz_ | but Cisco has a guide to configure that device on * |
| 19:23.02 | Tozz_ | so I think taht should work fine |
| 19:23.05 | javier_cintron | really?? |
| 19:23.05 | Tozz_ | https://supportforums.cisco.com/docs/DOC-9899/ |
| 19:25.00 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 19:25.19 | javier_cintron | hehe, I never came across this guide ;) |
| 19:25.42 | Tozz_ | first hit on google "asterisk spa8800" |
| 19:25.52 | leifmadsen | Tozz_: you cheated then |
| 19:25.58 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 19:25.59 | leifmadsen | Google?! pfffft! |
| 19:26.19 | leifmadsen | real men attempt random websites and search via web rings |
| 19:26.22 | Tozz_ | he must have used BING! |
| 19:26.42 | leifmadsen | Tozz_: I think Bing! returns the same results as Google (literally ;)) |
| 19:27.00 | Tozz_ | mm yes I've read that Bing! 'steals' search resulsts from google ;) |
| 19:27.20 | Tozz_ | on Bing! its result #4 ;) |
| 19:27.33 | leifmadsen | :) |
| 19:27.35 | leifmadsen | lol |
| 19:27.59 | Qwell | the other 3 are paid ads |
| 19:28.16 | leifmadsen | hawt |
| 19:29.07 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 19:49.55 | n3hxs | IF: provolone has quit (Remote host closed the connection) THEN who cut the cheese? |
| 19:54.40 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 19:59.30 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 20:03.49 | benngard | any have any plus or minus regarding Aastra 6757i? |
| 20:13.28 | leifmadsen | benngard: I think the over/under is -2.5 |
| 20:15.16 | benngard | u meen that i shouldnt buy them |
| 20:15.23 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 20:20.02 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 20:20.46 | leifmadsen | benngard: that's not what I said :) |
| 20:20.55 | leifmadsen | was trying to be funny |
| 20:20.59 | leifmadsen | I guess it got lost in the noise |
| 20:26.25 | n3hxs | too much jitter |
| 20:28.01 | skrusty | evening all |
| 20:29.21 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 20:31.10 | killown | do anyone here know if there is a good framework to speech to text?, I need get asterisk recorded call and translate it to text, sorry I don't know if my question is offtopic. |
| 20:32.37 | leifmadsen | killown: you could potentially pull the audio out from JACK, or there is a speech framework in Asterisk, but you'd have to develop the speech-to-text interface |
| 20:32.50 | beek | waves to leifmadsen |
| 20:32.53 | leifmadsen | o/ |
| 20:33.12 | killown | leifmadsen, this is my propose, develop the interface... |
| 20:34.12 | killown | leifmadsen on english calls I know there is would be easy to do, but in my language (portuguese) I think I have no alternative |
| 20:36.40 | leifmadsen | killown: well res_speech is a generic speech recognition API, but it's mostly for Lumenvox afaik |
| 20:36.53 | leifmadsen | it could potentially be a hook into Asterisk? I'm not sure. |
| 20:37.22 | leifmadsen | I guess it depends how much development you have to do? Speech recognition isn't a trivial matter. |
| 20:38.55 | killown | ok |
| 20:39.23 | leifmadsen | all the speech recognition would be done with software outside of Asterisk |
| 20:40.59 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 20:43.09 | *** join/#asterisk kreign (~kreign@64.179.176.6) |
| 20:44.07 | kreign | hi, I'm trying to set up a remote extension and I keep getting "chan_sip.c: Registration from '' failed for '192.168.33.127' - Device does not match ACL" for the remote phone. phone picks up its configuration but stays in the 'no service' state. anyone have an idea what kind of setting (on the phone or in asterisk) I'm looking for to fix this? |
| 20:44.13 | kreign | it's on a different subnet than the main phone network |
| 20:46.12 | pabelanger | kreign: your ACL settings in sip.conf. EG; allow / deny |
| 20:46.18 | Qwell | accept* |
| 20:46.29 | pabelanger | right ;) |
| 20:46.33 | Qwell | poorly named, heh |
| 20:46.42 | Qwell | err, no. hell. |
| 20:46.43 | Qwell | it's permit |
| 20:46.55 | wdoekes2 | hehe |
| 20:46.56 | Qwell | asterisk has broken my brain. |
| 20:48.03 | kreign | pabelanger, this is an asterisk device (made by ipitomy) so I don't actually have access to the sip.conf. :| what kind of variable is that option expecting? |
| 20:49.44 | Qwell | kreign: an IP address/mask |
| 20:49.52 | pabelanger | yup |
| 20:50.43 | kreign | Qwell, if the phone gets a 192.168.x.0/24 address, can I use a single 'permit' of something like 192.168.0.0/16? |
| 20:50.52 | kreign | or will it disallow? |
| 20:51.01 | Qwell | it accepts cidr |
| 20:52.56 | kreign | thanks |
| 20:53.47 | *** join/#asterisk pcangel (~yoink@www.vehiclestars.com) |
| 20:53.58 | kreign | can the ACL error relate to any other option? |
| 20:54.12 | kreign | eg. sip password, or the like |
| 20:54.29 | pabelanger | kreign: no, it would be a different error message |
| 20:54.32 | *** join/#asterisk manji (~manjiki@ppp-94-65-254-44.home.otenet.gr) |
| 20:54.37 | pcangel | Hi guys.. if I receive two Bridge events, that A is bridged to B, and B is bridged to C, and then channel B hangs up - is A then bridged to C, or do I end up getting a new event bridging them, or does A and C hang up? |
| 20:54.56 | kreign | pabelanger, thanks. |
| 20:55.37 | pabelanger | kreign: do you not have SSH or access to the source code of the device? |
| 20:55.56 | kreign | pabelanger, no. :| |
| 20:55.58 | pcangel | or does this never happen? ie conferencing happening via meetme events |
| 20:56.21 | pcangel | can a channel be bridged to more than one other channel? |
| 20:56.36 | kreign | pabelanger, vendor is ipitomy |
| 20:56.54 | *** join/#asterisk a_m_y (IceChat7@112.204.208.9) |
| 20:57.29 | kreign | though now that you mention it I should look into it to see if it's a GPL violation. |
| 20:57.35 | a_m_y | hi, may I know how to disable transfer extension to extension, am using 1.4.36. Thnx in advance |
| 20:57.42 | pabelanger | kreign: yes |
| 20:57.52 | kreign | pabelanger, though, IIRC, they'd only be entitled to give me the source, not a root login or anything like that... |
| 20:58.08 | pabelanger | kreign: correct |
| 20:58.31 | pabelanger | kreign: Trying booting to single user mode and seeing if you can change the files? |
| 20:58.44 | *** join/#asterisk iPod-nano (43ac523e@gateway/web/freenode/ip.67.172.82.62) |
| 20:59.29 | iPod-nano | I know I've done this before, and I know I've fixed it before, but I can't for the life of me remember how. When I get an incoming Gizmo5 call, and I hang up first, the other end doesn't hang up right away. |
| 21:01.00 | *** part/#asterisk diemos (~root@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 21:01.46 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 21:02.34 | *** join/#asterisk gemignani (~gemignani@201.86.212.46.static.gvt.net.br) |
| 21:04.09 | *** part/#asterisk gemignani (~gemignani@201.86.212.46.static.gvt.net.br) |
| 21:04.24 | iPod-nano | And Google was no help. |
| 21:07.05 | Tozz_ | try Bing! |
| 21:07.21 | drmessano-lt | Bing = Google :( |
| 21:07.40 | citywok | bing < google |
| 21:07.58 | drmessano-lt | Bing uses Google, so should you |
| 21:08.15 | Qwell | drmessano-lt: only because Google engineers use IE9. |
| 21:08.25 | Qwell | ie; Google engineers use IE, so should you. |
| 21:10.52 | kreign | pabelanger, unfortunately the board has no such access. |
| 21:11.32 | kreign | I'd have to destroy it to get at it... there are networking ports and a serial port (seemingly disabled?), and it looks like the machine was molded around the board. |
| 21:11.43 | kreign | molded = folded |
| 21:12.22 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 21:12.45 | kreign | pabelanger, the part of the error "Registration from ''" - should the '' have some sort of ID in there? |
| 21:13.12 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 21:13.33 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 21:14.44 | a_m_y | ~ebook |
| 21:14.50 | a_m_y | ebook |
| 21:14.53 | a_m_y | ^_^ |
| 21:14.58 | Qwell | ~newbook |
| 21:14.58 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 21:15.30 | a_m_y | ~transfer |
| 21:15.38 | a_m_y | ~sipnat |
| 21:15.38 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 21:15.55 | a_m_y | tnx |
| 21:21.57 | leifmadsen | A security release of Asterisk has just been released. For more information please see http://www.asterisk.org/node/51578 |
| 21:22.02 | *** join/#asterisk javier_cintron (~javier_ci@200-56-202-1-cable.cybercable.net.mx) |
| 21:22.15 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.4 (2010/02/21), 1.6.2.16.2 (2010/02/21), 1.4.39.2 (2010/02/21), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 21:23.30 | carrar | h4X0r |
| 21:34.32 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 21:38.12 | *** join/#asterisk Snif (~Fahd@41.248.128.226) |
| 21:38.23 | Snif | hello |
| 21:38.51 | Snif | i have asterisk server and i added some extentions and when i work with it i can only transfer 18 or 19 calls |
| 21:39.02 | Snif | how can i increase the channels limite? |
| 21:39.38 | leifmadsen | Asterisk imposes no limits on number of calls |
| 21:40.12 | Snif | so how come it wont pass 18 active channs ? |
| 21:40.28 | leifmadsen | hard to say without additional information |
| 21:40.31 | leifmadsen | is not here anymore |
| 21:40.45 | Snif | wich additional infos should i give you please |
| 21:41.09 | _Corey_ | what occurs when you reach the 19th or 20th call? |
| 21:41.11 | Kobaz | asterisk version, console logs and asterisk log files with core set debug 10 |
| 21:41.32 | Kobaz | well, first just with core set verbose 3 |
| 21:41.42 | Kobaz | and then if that's not enough info, then debug |
| 21:42.14 | titter | Anyone ever put multiple GotoIf's into one line? GotoIf($[$["${var}" = "x"]?result1 | $["${var}" = "y"]?result2]) ... this doesn't work, but maybe I am missing something obvious. |
| 21:42.31 | Kobaz | titter: i don't think that's supported |
| 21:42.38 | Kobaz | titter: apps take a very strict set of arguments |
| 21:42.46 | tzanger | I've been using asterisk for many years and I've never run across the need to do that |
| 21:42.52 | tzanger | I wouldn't try to be so cute with the code |
| 21:43.19 | Qwell | $[1?result1 | 0?result2] |
| 21:43.41 | Qwell | titter: that's what it ends up being evaluating as |
| 21:43.46 | Qwell | which just makes no sense |
| 21:44.19 | titter | Alright, didn't think so. Just curious more or less. |
| 21:44.20 | Kobaz | titter: that's not multple gotoifs.. that's multiple logical evaluation blocks |
| 21:44.30 | Snif | maybe there is a option on extentions for the limite of active calls for each extention? |
| 21:44.41 | Kobaz | $[] converts an expresssion into a boolean 0/1 |
| 21:44.58 | titter | Gotcha |
| 21:45.02 | Snif | cause i am talkin about when i use any extention from express talk for exemple and if i pass 19th active call i cant get more |
| 21:45.20 | Snif | when i try the 20th call it say packet lost |
| 21:45.31 | Kobaz | Snif: there are no configurable limits on calls to extensions... sip however has call-limit in the config file (and some other channel drivers have channel limits... which are being depricated) |
| 21:46.17 | citywok | Kobaz: you mean to an extensions.conf extension you create on your own? you could do it yourself pretty easily. |
| 21:46.32 | Snif | i create from the panel |
| 21:46.42 | Kobaz | citywok: you can write one yourself, sure... but nothing is built-in |
| 21:47.14 | _Corey_ | I think the call-limit has actually been removed 1.6.2+ |
| 21:47.16 | Snif | kobaz:i am not calling to extentions |
| 21:47.21 | Snif | i am calling international calls |
| 21:47.26 | Snif | i make transfers |
| 21:47.29 | Snif | between clients |
| 21:47.31 | Kobaz | you're still using an extension in the asterisk world |
| 21:47.38 | Snif | i can only let 18 peoples talking |
| 21:47.39 | Kobaz | 1234@internal |
| 21:47.42 | Kobaz | extension@context |
| 21:47.56 | *** join/#asterisk woglinde (~heinold@g225074060.adsl.alicedsl.de) |
| 21:48.20 | Snif | any idea! |
| 21:48.27 | Kobaz | we need more details |
| 21:49.19 | Kobaz | what devices are you calling from... what devices are you calling, are you using a line card, sip, iax, carrier pidgeon? you say packet loss... is your network causing the problems? |
| 21:49.40 | titter | Basically this was why I was looking to do that in one line, as when it hits the CLI it runs through all lines of syntax, and it's just getting a bit cumbersome. http://pastebin.com/nWPxD0wP |
| 21:49.52 | Kobaz | use AEL |
| 21:49.57 | woglinde | hm some one knows if the HST Saphir ISDN miniPCI works good enough with misdn? |
| 21:50.13 | titter | hmm |
| 21:50.32 | Kobaz | titter: AEL ftw |
| 21:50.42 | Kobaz | you'll never look back |
| 21:51.00 | Kobaz | http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
| 21:51.22 | titter | Sweet. Thanks again guys. |
| 21:51.25 | Kobaz | titter: scroll to the middle of tha page where it talks about conditionals |
| 21:51.29 | woglinde | the hst support didnt wrote me back for the fax option |
| 21:51.47 | Snif | i use sip |
| 21:51.55 | Snif | express talk software |
| 21:51.57 | Snif | for voip |
| 21:52.12 | Snif | i am calling cell phones |
| 21:52.20 | Snif | i call from my own computer |
| 21:52.28 | titter | Perfect ... time for some reading. |
| 21:52.28 | Kobaz | Snif: this has nothing to do with asterisk? |
| 21:52.40 | Snif | ? |
| 21:52.43 | Snif | what do u mean? |
| 21:52.46 | Kobaz | where does asterisk fit in |
| 21:52.52 | Kobaz | if you are using custom software |
| 21:53.02 | Snif | i have the panel |
| 21:53.08 | Snif | i monitor the calls |
| 21:53.15 | Kobaz | we don't support gui interfaces here |
| 21:53.23 | Snif | not only me who work on the company |
| 21:53.24 | Kobaz | what panel? asterisknow? |
| 21:53.25 | Snif | hmmm |
| 21:53.42 | Kobaz | the gui interfaces have their own support channels |
| 21:53.45 | p3nguin | AsteriskNOW is a distribution, not a panel. |
| 21:54.15 | Snif | i use the normal control panel |
| 21:54.16 | Snif | :s |
| 21:54.22 | Snif | there is diferent ones! |
| 21:54.23 | p3nguin | Oh, so you use vim. |
| 21:54.23 | Kobaz | we don't know what that is |
| 21:54.24 | _Corey_ | ah, that clears it up :( |
| 21:54.35 | p3nguin | The normal control panel is a text editor. |
| 21:54.39 | Kobaz | heh |
| 21:54.40 | Kobaz | emacs! |
| 21:54.47 | Kobaz | or... ed! |
| 21:54.56 | Snif | all what i see on the site link is https://xx.xx.xx.xx/auth |
| 21:54.56 | p3nguin | ee, joe, fifty others |
| 21:54.58 | Qwell | ed is the standard panel. |
| 21:55.14 | Kobaz | Snif: we have no idea what you see, or what software you are using |
| 21:55.23 | Snif | omg |
| 21:55.23 | Snif | :( |
| 21:55.25 | p3nguin | Need pics! |
| 21:55.29 | woglinde | hm uh now someone sells them for 1,99 on ebay |
| 21:55.36 | woglinde | I think for that price I will buy it |
| 21:55.49 | Snif | its digium asterisk |
| 21:55.57 | p3nguin | okay |
| 21:56.03 | Kobaz | Snif: what you are asking... is like walking to a mechanic, asking him to fix you car, without showing the cat |
| 21:56.05 | p3nguin | We know all about Asteisk. |
| 21:56.08 | Kobaz | *car |
| 21:56.19 | p3nguin | But we don't know anything about any other software which is not part of Asterisk. |
| 21:56.20 | Qwell | Kobaz: cat works just as well in that sentence :) |
| 21:56.23 | drmessano-lt | cat was more appropriate |
| 21:56.25 | Kobaz | haha |
| 21:56.34 | _Corey_ | Sniff: Look in the upper left hand corner of your browser... if it says FreePBX you might want to try #freepbx |
| 21:56.46 | Snif | no its not freepbx |
| 21:56.54 | Snif | i see digium asterisk |
| 21:57.00 | p3nguin | Show us. |
| 21:57.05 | p3nguin | Make us see it too. |
| 21:57.06 | Kobaz | printscreen |
| 21:57.07 | pcangel | I'm still very curious - I didn't get a reply last time I asked - can a channel be bridged to more than one other channel simultaniously, ie A <-> B A <-> C C <-> D and in that example, are D and A communicating with each other like a conference? |
| 21:57.09 | p3nguin | Take pictures. |
| 21:57.26 | drmessano-lt | Take a pic, SMS it to twitter, then follow us |
| 21:57.28 | _Corey_ | pcangel: Meetme() |
| 21:57.35 | Kobaz | drmessano-lt: haha |
| 21:57.45 | drmessano-lt | I use the name "Qwell" on twitter |
| 21:57.50 | Snif | ok |
| 21:57.55 | pcangel | I know about Meetme, that's not why I want to know- I just need it know if the circumstance I listed is possible or if it can't occur |
| 21:58.09 | pcangel | I'm writing a C++ app that turns all of the AMI information in to a series of structures for use with my company's call centre management app |
| 21:58.26 | Kobaz | drmessano-lt: when did you become qwell? |
| 21:58.35 | Qwell | when he bought the trademark |
| 21:58.39 | drmessano-lt | Kobaz, mind your cat |
| 21:58.46 | pcangel | and I am trying to figure out if I should make my bridge code a large linked list of Channel1/Channel2 or if I should just put a Bridged_to variable in my channel structure |
| 21:58.50 | Kobaz | hmm |
| 21:59.17 | pcangel | I don't know how people might use it in the future, but it'd save processing time and result in cleaner code if I'm able to just have a Bridged_to member of class CHANNEL |
| 21:59.19 | _Corey_ | pcangel: I prefer to parse the 'core show channels' output via AMI or whatever |
| 21:59.36 | pcangel | I agree, but I'm reading the events |
| 21:59.51 | pcangel | that's how I initialize the information when this server-app first starts up |
| 22:00.15 | *** join/#asterisk pagec (~chatzilla@96.57.210.34) |
| 22:00.30 | Snif | http://www.ayatco.com/aster.png |
| 22:00.32 | Snif | here is it |
| 22:00.47 | Qwell | what the crap is that? |
| 22:00.56 | Snif | the panel i use |
| 22:00.57 | _Corey_ | seriously... ? |
| 22:01.06 | Snif | to control my asterisk |
| 22:01.13 | pagec | i recently upgraded my polycom phones to 3.3.1, and now when calls come into them (from asterisk), instead of another line light lighting up the users have to scroll down and use the "answer" softkey, anyone else have this problem? |
| 22:01.16 | p3nguin | I've never seen such a thing. |
| 22:01.17 | Kobaz | where did you get it? |
| 22:01.19 | Qwell | click the "PBX Info/License" link at the bottom left |
| 22:01.22 | Qwell | show us that page |
| 22:01.23 | Snif | ? |
| 22:01.25 | *** join/#asterisk isrl (~isrl@IGLD-84-228-224-139.inter.net.il) |
| 22:01.25 | titter | The hell is that? |
| 22:01.36 | Snif | ok |
| 22:01.46 | isrl | Hi |
| 22:01.58 | p3nguin | I believe the phrase is, "What the hell is that?" |
| 22:01.59 | _Corey_ | It's PHP, whatever it is |
| 22:02.29 | Snif | there is pbx version |
| 22:02.34 | Snif | and warranty |
| 22:02.35 | Snif | ...etc |
| 22:02.46 | _Corey_ | pagec: You need to look at the calls per line key option |
| 22:02.47 | p3nguin | Yeah... click it and take a pic. |
| 22:03.00 | Snif | ok |
| 22:03.21 | Kobaz | Snif: did you buy this? |
| 22:03.25 | isrl | madplay doesnt work in 1.8.2 1.8.3 only if noload res timing dahdi |
| 22:03.47 | pagec | _Corey_: ok ty |
| 22:03.48 | titter | p3nguin: Depends. |
| 22:03.58 | isrl | but if i turn that off i get deadlocks couple times a day |
| 22:04.10 | isrl | any help? |
| 22:05.07 | titter | p3nguin: You would want to use brackets to quote what I said, as I literally said what I typed. However to clarify the expression and tone of my reaction to those reading it [What] would be more proper lol. |
| 22:06.23 | Snif | http://www.ayatco.com/aster.png |
| 22:06.31 | Snif | check it out |
| 22:07.05 | _Corey_ | Snif: Did you download this software from somewhere or did you buy this? |
| 22:07.15 | Snif | i boughout it |
| 22:07.19 | Qwell | from who? |
| 22:07.20 | Snif | why? |
| 22:07.30 | Snif | from someone |
| 22:07.32 | _Corey_ | We've never seen it before |
| 22:07.35 | Snif | :o |
| 22:07.38 | Snif | why no ? |
| 22:07.41 | Snif | its asterisk |
| 22:08.04 | _Corey_ | No, it's not |
| 22:08.04 | Snif | but u see digium asterisk ! |
| 22:08.17 | _Corey_ | There may be Asterisk on the hardware somewhere but that doesn't make "it" Asterisk |
| 22:08.23 | Qwell | From where did you buy it? |
| 22:08.30 | Snif | online |
| 22:08.34 | Qwell | WHERE? |
| 22:08.42 | Snif | someone sold it to me |
| 22:08.48 | Snif | just a friend |
| 22:08.49 | Qwell | ... |
| 22:08.54 | _Corey_ | Snif: It would help us to understand if you could point to a website or something |
| 22:09.06 | titter | Then contact your friend to fix the problem with HIS pbx |
| 22:09.15 | Snif | there is no really a website |
| 22:09.25 | Snif | cause its only someone who sells that |
| 22:09.34 | Snif | is this could be what providers use ? |
| 22:09.42 | leifmadsen | *coughscamcough* |
| 22:09.43 | _Corey_ | definitely not |
| 22:09.47 | Qwell | No provider in their right mind would use garbage like this. |
| 22:09.48 | Snif | cause i see some options where to add provider |
| 22:10.02 | Qwell | Throw it away, get a real PBX. |
| 22:10.12 | Snif | voip provider |
| 22:10.14 | Qwell | Probably didn't cost more than what, $40? |
| 22:10.22 | Snif | lool |
| 22:10.25 | Snif | u mean it? |
| 22:10.34 | Snif | u mean this is only a shit? |
| 22:10.37 | Qwell | You're getting a load average of 6 with 18 calls. |
| 22:10.40 | Qwell | Yes. |
| 22:11.01 | Kobaz | Snif: if you paid more than $5 for it, you should give it back |
| 22:11.18 | Snif | :( |
| 22:11.21 | Kobaz | even thirdland would be better |
| 22:11.24 | Kobaz | thirdlane |
| 22:11.28 | Kobaz | man i can't type today |
| 22:11.35 | p3nguin | "u mean this is only a shit?" That needs to go on a T-shirt like the other saying. |
| 22:11.45 | _Corey_ | lol |
| 22:11.59 | p3nguin | Some of you remember the other T-shirt, I'm sure. |
| 22:12.16 | Snif | there is no other option where can i find something usual for u ? |
| 22:12.27 | isrl | where could i get help on something that might be a bug? |
| 22:12.40 | Qwell | Snif: No, you won't get any help for that here. |
| 22:12.55 | Kobaz | Snif: it's an interface written by who-the-hell-knows, using code that we know nothing about |
| 22:13.11 | _Corey_ | Snif: Take a look at FreePBX or better yet Switchvox |
| 22:13.12 | Snif | so what is ur advice? |
| 22:13.18 | Qwell | <Qwell> Throw it away, get a real PBX. |
| 22:13.20 | Kobaz | Snif: get something that we can support |
| 22:13.22 | Snif | freepbx is good? |
| 22:13.27 | Kobaz | and it's free |
| 22:13.47 | Snif | what about tribox? |
| 22:13.55 | Kobaz | that's fine too |
| 22:14.01 | Qwell | glares at Kobaz |
| 22:14.03 | Kobaz | but you'll need to get support in #trixbox |
| 22:14.06 | _Corey_ | lol |
| 22:14.07 | Qwell | Kobaz: I don't want to see him again in 3 days. |
| 22:14.08 | p3nguin | Good? No... FreePBX is not good, but it is free. |
| 22:14.19 | Tozz_ | which makes it perfect for most! |
| 22:14.31 | p3nguin | Trixbox is worse because there's no support for it. |
| 22:14.34 | Kobaz | p3nguin: he wants something a complete newb can use, so it's better than what he has now, because he can get help with it |
| 22:14.35 | _Corey_ | shushes p3nguin |
| 22:14.39 | Snif | Qwell:i will leave foreven not only for 3 days |
| 22:14.40 | Snif | :s |
| 22:15.02 | Kobaz | p3nguin: there was some people around in #trixbox the last time i looked |
| 22:15.05 | Kobaz | but anyway |
| 22:15.07 | Snif | thanks guys |
| 22:15.17 | Kobaz | you'll want switchvox |
| 22:15.26 | p3nguin | How much does that cost? |
| 22:15.35 | Kobaz | they have a free home version |
| 22:15.40 | p3nguin | oh |
| 22:15.54 | p3nguin | is interested in trying something new. |
| 22:16.06 | p3nguin | Not that I don't do just fine with vim. |
| 22:16.08 | Snif | sounds good |
| 22:16.15 | Kobaz | the dashboard stuff looks pretty cool in the new switchvoxes |
| 22:16.27 | Kobaz | from one of the talks i went to at astricon |
| 22:16.48 | _Corey_ | There's an iPhone app.. |
| 22:16.54 | Kobaz | Snif: if you want something free, freepbx |
| 22:16.56 | _Corey_ | i mean, what else do you possibly need |
| 22:17.08 | Snif | i will try frreepbx |
| 22:17.32 | Snif | anyone know about voip reseller? |
| 22:17.42 | Snif | or any channel support that kind of informations? |
| 22:17.48 | Kobaz | #freepbx |
| 22:17.53 | Kobaz | voip... |
| 22:17.56 | Kobaz | ~itsp |
| 22:17.56 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 22:18.02 | Kobaz | ~itsp-us |
| 22:18.12 | Qwell | isrl: never do that again |
| 22:18.23 | isrl | sorry |
| 22:18.30 | Kobaz | ~itsp-us |
| 22:18.31 | isrl | where could i get help on something that might be a bug? |
| 22:18.42 | Kobaz | pokes infobot |
| 22:18.49 | Snif | itsp-us is a site? |
| 22:18.55 | Kobaz | oh whoops |
| 22:18.56 | Kobaz | ~itsplist-us |
| 22:18.56 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 22:20.10 | isrl | madplay doesnt work in 1.8.2 1.8.3 only if noload res timing dahdi |
| 22:20.22 | Kobaz | isrl: you already asked that |
| 22:20.37 | Kobaz | isrl: pastebin up your console logs and anything else you can get |
| 22:20.51 | Snif | they sell the numbers or its free? |
| 22:20.58 | Snif | i mean teliax |
| 22:21.04 | isrl | ok |
| 22:21.50 | Kobaz | of course it's not free |
| 22:21.54 | isrl | on the console it just says that its plaing but i dont hear anything |
| 22:22.06 | isrl | *playing |
| 22:22.45 | Kobaz | make sure you have valid music, and check debug output from madplay |
| 22:23.17 | Snif | anyone hear about goldex-telecom.com? |
| 22:23.27 | Snif | i wont be victime of scam |
| 22:23.27 | Snif | :s |
| 22:23.34 | Snif | they have free premium numbers |
| 22:23.47 | isrl | if i put noload=>res_timing_dahdi.so in modules then it works |
| 22:23.51 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 22:24.31 | Kobaz | ~cheap |
| 22:24.31 | infobot | cheap is, like, a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 22:25.09 | isrl | it might be the same bug as the parking one but i upgraded to latest rc where that was fixed and this is still a problem |
| 22:25.16 | Kobaz | Snif: the number might be free... but you have to pay to actually use it |
| 22:25.29 | Kobaz | Snif: at least there's a hot girl on the main page... right? |
| 22:25.47 | Snif | yea lol |
| 22:25.57 | Snif | how can i know if they are not scammers! |
| 22:26.02 | Snif | they dont have any forum |
| 22:26.08 | Tozz_ | then their scammers! |
| 22:26.08 | Snif | i am confused |
| 22:26.13 | Tozz_ | any sane ISP has a forum! |
| 22:26.14 | Tozz_ | doh! |
| 22:26.23 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 22:26.30 | Kobaz | a real isp wouldn't have a forum |
| 22:26.35 | *** join/#asterisk NetAtWorld (~support@ool-45794838.dyn.optonline.net) |
| 22:26.39 | Kobaz | Snif: the 'about us' page is blank |
| 22:26.43 | Kobaz | Snif: find someone else |
| 22:27.53 | Snif | check contact us |
| 22:28.04 | Snif | they pretende to be in hong kong |
| 22:29.03 | Qwell | yeah they seem legit |
| 22:29.36 | Snif | how did u know ? |
| 22:30.39 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:31.15 | Kobaz | he's being sarcastic |
| 22:32.27 | Snif | ah ok |
| 22:32.30 | *** join/#asterisk ectospasm (ectospasm@188.72.223.139) |
| 22:32.56 | Snif | wich site do u advice me guys ? |
| 22:33.03 | Kobaz | ~itsp |
| 22:33.03 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 22:33.20 | Kobaz | one of those.. well... one of those |
| 22:33.26 | Kobaz | from the us/ca lists |
| 22:33.44 | Snif | i dont have credit card to register with them |
| 22:33.44 | Snif | :s |
| 22:33.48 | Kobaz | some of them do international as well |
| 22:34.37 | Kobaz | are you 12? |
| 22:34.52 | Snif | lol |
| 22:34.53 | Snif | no |
| 22:35.03 | Kobaz | get a credit card then |
| 22:35.04 | Snif | i am from morocco |
| 22:35.06 | Snif | thats why |
| 22:35.11 | Snif | i dont have international vsa |
| 22:35.13 | Snif | visa |
| 22:35.13 | Kobaz | oh, hmm |
| 22:35.16 | Snif | only national |
| 22:35.29 | Kobaz | maybe ask in #banking |
| 22:35.43 | Kobaz | this is going out of the scope of #asterisk |
| 22:36.05 | Snif | oh ok |
| 22:36.14 | Snif | sorry to break the channel's rules |
| 22:36.15 | Snif | ^^ |
| 22:36.36 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 22:36.39 | Kobaz | not really rules, but i'm just getting tired of all the random questions that aren't asterisk related |
| 22:36.41 | nestAr | can you do MOH from dsp? |
| 22:36.55 | Kobaz | but ideally you should stay on topic |
| 22:37.18 | Snif | its just bcos i dont want to be a victim of scam |
| 22:37.24 | nestAr | seems like a really bad idea, but i have a feeling my boss is going to ask for it. |
| 22:37.27 | Snif | as u can see i have a bad version of asterisk |
| 22:37.27 | Snif | :s |
| 22:37.39 | Snif | thanks for u |
| 22:37.40 | Kobaz | ~itsplist-us |
| 22:37.40 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 22:37.41 | Snif | for all |
| 22:37.46 | Kobaz | any of those are legit companies |
| 22:37.58 | Kobaz | as well as voip.ms also |
| 22:38.09 | Snif | that one shows nothing |
| 22:38.11 | Snif | blank page |
| 22:38.27 | Kobaz | voip.ms has UK servers |
| 22:38.44 | Kobaz | Snif: maybe your internet provider is blocking them |
| 22:38.58 | Snif | yes |
| 22:39.00 | Snif | u are right |
| 22:39.04 | Snif | i pasted it to a friend |
| 22:39.08 | Snif | its opening with him |
| 22:39.29 | _Corey_ | Snif: Does your government filter internet access? |
| 22:39.46 | Snif | not really |
| 22:39.53 | Qwell | read: yes |
| 22:39.57 | Kobaz | heh |
| 22:40.04 | Snif | but maybe the voip.ms |
| 22:40.15 | Snif | maybe my ip is on the blacklist |
| 22:40.15 | Snif | lol |
| 22:40.19 | *** join/#asterisk tash (~Tommy@ks-76-7-1-196.sta.embarqhsd.net) |
| 22:40.32 | Kobaz | goes back to under his rock |
| 22:40.44 | tash | can you pass a variable from an agi script back to the dial plan? |
| 22:40.44 | *** join/#asterisk ectospasm (ectospasm@188.72.223.139) |
| 22:41.26 | nestAr | tash: yes.. |
| 22:41.58 | tash | I think that is what I want to do anyway :P |
| 22:42.15 | killown | does anyone here use Starpy http://www.vrplumber.com/programming/starpy/ to develop asterisk apps, I'd like to know if this thing has a good support to provides access asterisk manager interface? |
| 22:43.51 | tash | nestAr: I basically want to limit the number of call attempts made to a phone number if they don't answer their phone. That way I'm not trying to call over and over again. So, was thinking I could update the DB for each call attempt, and when that record reached 5 attempts, stop calling...but that'd involve doing an 'if' in the dial plan ... idk, I might be thinking too much |
| 22:44.08 | Snif | if i change my dns its gonna alow me to access voip.ms ? |
| 22:48.39 | isrl | kobaz: could you give me a idea of how i could debug that problem the console and logs say that its starting music class default and madplay is showing as playing |
| 22:49.04 | *** join/#asterisk coppice (~chatzilla@m121-203-224-167.smartone-vodafone.com) |
| 22:49.04 | isrl | but no audio |
| 22:49.13 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:52.05 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
| 22:53.13 | isrl | this is the deadlock i get wen usin the pthread timer http://pastebin.com/raw.php?i=6wPXHPcz |
| 22:53.25 | Kobaz | isrl: you need to verify that madplay is actually sending audio, and asterisk is recieving it |
| 22:54.57 | tash | can someone advise? -> in my dialplan I am expecting a user to press 1 to confirm they heard the call. If they don't, I am calling them back in 5 minutes. I want to stop calling them after 5 attempts where they haven't pressed 1 to confirm they heard the call (basically call goes to voicemail and a digit is never pressed). Any advice on how to check a record in my database and stop making the calls to them if the record is 5? |
| 22:55.14 | isrl | thats my question how could i debug that |
| 22:55.24 | p3nguin | What could be some reasons that a fresh build of asterisk 1.8.2.4 segfaults a second after running it? |
| 22:55.34 | _Corey_ | ouch |
| 22:55.41 | _Corey_ | I was just going to download it... hmm |
| 22:55.41 | isrl | because madplay says playing |
| 22:55.46 | WIMPy | tash: How to you trigger the call back? |
| 22:55.47 | Kobaz | isrl: not sure exactly |
| 22:56.05 | fauxalliance | p3nguin, two idential jabber registrations did it for me. |
| 22:56.54 | p3nguin | _corey_: I figure it has something to do with this system as it is configured, but I was hoping to get some ideas of what to look at to solve it. |
| 22:57.14 | tash | WIMPy: if they don't press 1, I move the call file to a tmp dir, give it a timestamp that is 5 minutes from then, and then a cron runs every 1 minute and moves .call's from that dir to asterisk's outgoing dir |
| 22:57.17 | _Corey_ | p3nguin: Did you compile it so that it will core dump? |
| 22:57.22 | *** part/#asterisk woglinde (~heinold@g225074060.adsl.alicedsl.de) |
| 22:57.53 | WIMPy | p3nguin: No indication if you start with console? Otherwise look at the core dump. |
| 22:57.59 | p3nguin | _corey_: If that's a special option, no. I guess I'll do that pretty soon if I can't find something causing a problem. |
| 22:58.07 | WIMPy | tash: Then just put a variable with a counter in to the call file. |
| 22:58.15 | p3nguin | wimpy: Nothing jumps out at me. |
| 22:58.22 | _Corey_ | p3nguin: Usually when we're testing, we add DONT_OPTIMIZE and DEBUG_THREADS |
| 22:59.42 | p3nguin | Good idea. If I can ever figure out what's causing the problem and get a good build, I'm ready to start testing 1.8 to prepare for migration from 1.4. |
| 23:00.15 | tash | WIMPy: like exten => mycaller,n,Set(__counter=${0}) or something? I'm really new to most of this, so I may not have the right idea or even the right syntax |
| 23:00.26 | tash | then how would I check the var? |
| 23:01.34 | WIMPy | There's a 'setvar' in call files. |
| 23:01.47 | p3nguin | This seems problematic: format_mp3.c:38:24: fatal error: mp3/mpg123.h: No such file or directory |
| 23:02.12 | WIMPy | And then you can check in your dialplan with ExecIf or GotoIf. |
| 23:02.24 | tash | WIMPy: ah yes, I see ... I am actually setting a var already in my call :) |
| 23:03.07 | tash | WIMPy: so, SetVar: COUNTER=0 could be the var I set in the call file, right? Then, on the retry attempt, I could update that by 1 or something ...? |
| 23:03.41 | WIMPy | That's the idea. |
| 23:03.46 | tash | I presume there is a way to use ExecIf to check a variable in the call file .. I will read the documentation. But, am I on the right track? |
| 23:03.52 | tash | ok, you just answered that :P thanks WIMPy |
| 23:04.08 | WIMPy | Not in the call file. On your Dialplan. |
| 23:04.33 | tash | Yah, I meant, a way to look in the call file and look for that string or something? |
| 23:05.08 | WIMPy | You check that variable. |
| 23:05.21 | tash | ah, makes sense |
| 23:05.52 | tash | and, say it checks it and it is currently at 5 and I don't want to make any more calls to that number I can tell the dial plan to then, say, Hangup? |
| 23:06.46 | WIMPy | Yes, or just don't schedule a new call. |
| 23:07.36 | tash | ok, well I can figure that out ... thank you very much for your input |
| 23:08.08 | javier_cintron | Hi there, in asterisk 1.8, what user owns asterisk daemon, isn't root right? |
| 23:09.00 | WIMPy | javier_cintron: The one you configured or the one it got started from. |
| 23:10.11 | p3nguin | javier_cintron: It's root if you started it as root. "ps -C asterisk u" will show you |
| 23:10.20 | tash | can't you do a ps auxf and grep for asterisk to see who owns it? |
| 23:10.36 | p3nguin | Or just do what I said and forget about grepping. |
| 23:10.37 | tash | p3nguin: cool, didn't know that |
| 23:12.35 | javier_cintron | ok, I compiled it usig root, so root owns it. if I wanted to use a different one, like asterisk for example, Would It be enough to compile it using this asterisk user?? |
| 23:13.29 | WIMPy | Take a look at asterisk.conf. |
| 23:13.43 | javier_cintron | ok thank you very much |
| 23:13.46 | WIMPy | Otherwise start it from that users. |
| 23:13.53 | WIMPy | -s |
| 23:14.15 | p3nguin | I don't remember having to run contrib/scripts/get_mp3_source.sh in 1.8.2.3 ... is that new today? |
| 23:14.43 | WIMPy | Not that new. |
| 23:14.45 | p3nguin | javier_cintron: It doesn't matter who compiled it, it matters how you run it. |
| 23:15.11 | isrl | is there anyway getting past the first song using format mp3? |
| 23:15.19 | WIMPy | I think it came when the addons were integrated. |
| 23:15.28 | p3nguin | Maybe I didn't enable the mp3 thing when I compiled 1.8.2.3. |
| 23:15.37 | p3nguin | I was certain I did. |
| 23:19.34 | javier_cintron | p3nguin: all right, I am seing in asterisk.conf a couple of parameters called: runuser and rungroup |
| 23:22.39 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:27.50 | titter | Anyone have some further reading material on AEL than what is on the voip-info page? |
| 23:27.54 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 23:28.55 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
| 23:30.39 | p3nguin | javier_cintron: Set those to asterisk and asterisk. Make sure you have the asterisk user and asterisk group on your system. |
| 23:31.38 | p3nguin | I do it like this: groupadd -g 5060 asterisk && useradd -c "Asterisk daemon" -u 5060 -d /var/lib/asterisk -g asterisk -s /bin/false asterisk |
| 23:34.44 | javier_cintron | p3nguin: let me try it, thanks :) |
| 23:36.22 | hardwire | asterisk asterisk asterisk asterisk |
| 23:36.26 | hardwire | asterisk asterisk asterisk |
| 23:36.29 | hardwire | asterisk asterisk asterisk asterisk asterisk |
| 23:36.31 | hardwire | asterisk! |
| 23:37.05 | javier_cintron | * * * * * * |
| 23:37.23 | p3nguin | wimpy: Here's what happens next: Unable to access the running directory (Permission denied). Changing to '/' for compatibility. |
| 23:38.17 | p3nguin | Then it loads a bunch of stuff, followed by seg fault. |
| 23:40.55 | p3nguin | I also don't see any core dump. |
| 23:42.15 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:42.44 | p3nguin | Changing autoload to no allows it to start without seg fault, so it's a problem in some random module. |
| 23:44.39 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 23:49.37 | javier_cintron | is it a good idea to run: chown -R asterisk:asterisk /usr/lib/asterisk/modules?, I am trying to run asterisk with a asterisk user |
| 23:51.06 | *** join/#asterisk dtoronto (~dtoronto@173-14-233-70-Utah.hfc.comcastbusiness.net) |
| 23:52.05 | WIMPy | p3nguin: Did you start it nofork and with high verbose and debug? |
| 23:52.25 | p3nguin | started with -vvvv |
| 23:53.15 | p3nguin | javier_cintron: chown -R asterisk:asterisk /var/run/asterisk /var/log/asterisk /var/spool/asterisk /var/lib/asterisk |
| 23:53.22 | WIMPy | If you're lucky you will se which module acts up. Or the last one that succeeded. In that case only the core will help. Or another system with the same modules. |
| 23:53.47 | *** join/#asterisk aarondev (~mod@farang.androids.name) |
| 23:53.51 | WIMPy | Nah, at least -cgvvvdddddddd |
| 23:54.09 | p3nguin | I've found out that I'm not quite ready for 1.8 anyway. chan_sccp-b won't compile against it. |
| 23:54.33 | p3nguin | I'm still interested in testing builds and getting asterisk running, though. |
| 23:55.41 | WIMPy | Create a patch. |
| 23:55.57 | *** join/#asterisk [canniballllera] (~cannibale@201-2-229-166.fnsce703.dsl.brasiltelecom.net.br) |
| 23:57.51 | dtoronto | Hi all, I have a distributed call center setup, asterisk1 = queue server, asterisk2 = extension register server. trying to route dynamic extensions from the asterisk1 to the respective extensions registered on the register servers. I am using XMPP for distributed device state, which is working. Problem is that I cannot get the queue to route calls to dynamic extensions. I have tried creating sip extensions on asterisk1 |
| 23:57.52 | dtoronto | <PROTECTED> |
| 23:58.46 | *** part/#asterisk [canniballllera] (~cannibale@201-2-229-166.fnsce703.dsl.brasiltelecom.net.br) |
