| 00:10.36 | *** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net) |
| 00:11.53 | ipengineer | Does anyone know what happens if I start MixMonitor on a channel and then later on in the flow set it again as it goes through my dialplan? Will the seconds one be ignored? |
| 00:21.27 | *** join/#asterisk brittel (~root@542713e4.ftth.concepts.nl) |
| 00:23.38 | brittel | does the dahdi driver handle q931? |
| 00:23.59 | WIMPy | No. libpri. |
| 00:24.56 | bsdice | brittel isn't that libpri's job? |
| 00:25.55 | brittel | ah ok |
| 00:26.00 | *** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net) |
| 00:35.03 | bsdice | brittel recommend to get rid of isdn anyhow... if you can |
| 00:37.48 | brittel | I'm using GSM |
| 00:38.14 | brittel | I want to dump the Q.931 signalling from the d-channel |
| 00:39.31 | brittel | much harden then i initially though |
| 00:39.33 | brittel | t* |
| 00:42.26 | WIMPy | GSM is not exactely Q.931. But it's based upon it. |
| 00:42.33 | brittel | I'm aware |
| 00:42.56 | brittel | Could it be possible that the whole signalling component is part of the baseband modem itself? |
| 00:43.16 | WIMPy | yes |
| 00:43.21 | brittel | the openvox card + drivers i have solely communicate with the modem through AT commands |
| 00:43.33 | WIMPy | It seems very likely. |
| 00:43.57 | WIMPy | Ok, then they work just like any "modem" or USB stick. |
| 00:43.57 | brittel | There a driver for 921 (d-channel) |
| 00:44.28 | WIMPy | That doesn't seem to make much sense. |
| 00:45.28 | bsdice | igh |
| 00:45.30 | bsdice | ugh |
| 00:45.31 | brittel | there is a full framing implementation in the driver (base.c) |
| 00:45.33 | bsdice | no idea :) |
| 00:47.16 | brittel | OpenVox GSM/CDMA card Dahdi driver |
| 00:49.17 | brittel | it implements the GSM 07.10 multiplexing protocol btw |
| 00:50.54 | brittel | hmm this is just a modem communication protocol |
| 00:51.12 | brittel | For example, you can send or receive data on the first channel, |
| 00:51.13 | brittel | while the other two channels are free to control the GSM/GPRS engine with AT commands. |
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| 02:17.52 | NivaldoMJr | Hi, Does ano |
| 02:18.29 | NivaldoMJr | Hi, Does anyone has tested asterisk 12 using pjsip and SRTP? |
| 02:20.54 | NivaldoMJr | I have two Yealink IP Phones that are registered. Calls using RTP are completed but we have no audio on both phones. with SRTP the call is not processed, i saw a not acceptable here. |
| 02:24.47 | NivaldoMJr | I need some help with this tests. If anyone has interest and has already tested the new chan_pjsip, contact me please. I will pay for this support. |
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| 04:26.22 | linocisco | in asterisk setup, calling from one extension to another, I would like to ring quickly without any delay , without any need to press # at the end of extension no. how could we do that? |
| 04:28.21 | spicyramen_ | you may have an overlap dialplan |
| 04:28.27 | spicyramen_ | in your phone or in asterisk |
| 04:28.35 | [TK]D-Fender | It's your phone.... |
| 04:28.45 | spicyramen_ | example: Asterisk extension 101 and Route pattern 1XXXXXXXXX |
| 04:29.03 | [TK]D-Fender | Asterisk doesn't make you wait to dail |
| 04:29.06 | spicyramen_ | some phones (cisco, GRandstream have internal dial plans you may take a look) |
| 04:29.14 | [TK]D-Fender | unless you're on a DAHDI analog FXS channel |
| 04:29.27 | [TK]D-Fender | everything else = phone |
| 04:30.17 | [TK]D-Fender | Actually, should revise that... technically any VoIP phone. |
| 04:30.38 | [TK]D-Fender | a BRI phone would match&go as well. |
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| 05:08.11 | linocisco | [TK]D-Fender, I dont understand well about internal dial plan on some phones |
| 05:09.00 | [TK]D-Fender | Things work better when you don't use generic terms like "some phones" and actually look at specific models |
| 05:09.28 | linocisco | spicyramen_ said some phones (cisco, GRandstream have internal dial plans you may take a look. |
| 05:10.44 | [TK]D-Fender | Almost every SIP phone has their own |
| 05:11.00 | [TK]D-Fender | I'm pretty sure I don't care about the models you DON'T have. |
| 05:24.09 | *** join/#asterisk ruben23 (~OpenDial@112.198.90.99) |
| 05:27.45 | ruben23 | hi anyone can help my asterisk is being attacked ---> http://pastebin.com/Jw2VEtVQ <---------- but the ip only appearing xxx.xxx.xx.xx is the IP of teh asterisk itself, i also deleted the extension 3086 but still flooding my asterisk CLI...please help someone..?? |
| 05:31.48 | [TK]D-Fender | <PROTECTED> |
| 05:31.58 | [TK]D-Fender | you aren't looking at the full packets |
| 05:32.04 | [TK]D-Fender | go get the actual inbound IP. |
| 05:32.11 | [TK]D-Fender | Older *'s don't have it on that warning line |
| 05:36.23 | ruben23 | [TK]D-Fender: please help how do i check..? |
| 05:36.29 | [TK]D-Fender | SIP DEBUG <- |
| 05:37.31 | ruben23 | its so fat how where do i cehck on this part..? |
| 05:37.35 | ruben23 | fast* |
| 05:39.08 | [TK]D-Fender | captue a pile. You'll see the IP. |
| 05:39.13 | [TK]D-Fender | This is not difficult |
| 05:39.19 | [TK]D-Fender | capture a pile* |
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| 07:03.57 | BarthezZ | does anyone has any idea what the impact on an operational asterisk system would be to change the time (which didn't had ntp configured and now has half an hour offset) |
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| 08:40.55 | michael_work | for local channel on 1.8 sip_cause is saved in HASH() while for 11 we have HANGUPCAUSE and apperantly it's not saved or i did not find way to find it in HASH |
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| 09:26.17 | aum | hi - new to asterisk, trying to connect to asterisk server via qutecom, getting "error while discovering network for SIP" |
| 09:26.36 | aum | can anyone offer some leads for troubleshooting this? |
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| 09:56.53 | Ice_Strike | Does real time call recording work well shared storage or recommended to use local disks? |
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| 09:57.46 | jwww | Hello. |
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| 14:00.58 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 14:08.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.13 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 14:09.58 | dym | Wonderful day, everyone! |
| 14:12.06 | Katty | indeed :> |
| 14:14.38 | Chainsaw | A very good day. I have a purchase order signed and permission to spend a little more still :) |
| 14:15.14 | Katty | cheers |
| 14:15.36 | Chainsaw | However, just a heads-up for the Polycom users... UCS 4.0.6 breaks TCP(only) support again. Stick to 4.0.5 rev C if you care about these things. |
| 14:15.36 | Ice_Strike | Have anyone used asterisk live recording on the Shared Storage instead of local disks? |
| 14:15.45 | Ice_Strike | It be interesting how does it perform. |
| 14:16.08 | Chainsaw | Ice_Strike: I think Greenlight uses trickery like that. |
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| 14:16.44 | dym | Ice_Strike: UCS as in Univention Corporate Server? |
| 14:20.06 | file | I'd do it in a ramdisk and move it to shared storage afterwards |
| 14:23.40 | Ice_Strike | @file ramdisk is too risky. I would store it in SSD local disk and then move it to shared storage afterwards |
| 14:23.55 | Ice_Strike | How do you move it to shared storage afterwards? |
| 14:24.28 | [TK]D-Fender | Ever copied a file before? |
| 14:24.31 | file | I would hope that if the system is that important it would be on a backup battery source, so it could be cleanly shut down |
| 14:24.38 | [TK]D-Fender | Kinda like that only you'd delete the source file after ;) |
| 14:25.47 | Ice_Strike | @file Sometime you can't trust backup battery :) |
| 14:26.24 | Ice_Strike | @file cleanly shut down will still clear the RAM |
| 14:26.32 | Ice_Strike | so make no difference. |
| 14:26.48 | file | all depends on your architecture. |
| 14:27.12 | Ice_Strike | Unless you have a script executing the script before shutdown :) |
| 14:27.56 | Ice_Strike | Im thinking getting second host ESXi and a shared storage. |
| 14:28.08 | Ice_Strike | the Vm's will be stored on the shared storage |
| 14:28.22 | Ice_Strike | so if a host goes down, I can run a VM from other host. |
| 14:28.24 | Ice_Strike | pretty cool. |
| 14:29.30 | davlefou | Hi, did you have already install opus under asterisk? |
| 14:31.18 | rrittgarn | Ice_Strike: You just described my production environment. it works well |
| 14:33.22 | Ice_Strike | rrittgarn which ESXi you using |
| 14:33.39 | rrittgarn | VCenter 5.0 |
| 14:33.42 | rrittgarn | (Vmware) |
| 14:34.05 | rrittgarn | just haven't upgraded to 5.5 on all my boxes |
| 14:36.09 | Ice_Strike | VMware vSphere Essentials Plus? |
| 14:36.27 | Ice_Strike | with vMotion |
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| 15:34.19 | drmessano | HP sucks |
| 15:34.26 | drmessano | Relevance: exten => 269,1,System(lp /var/log/asterisk/full) |
| 15:34.36 | drmessano | keeps it on topic, kinda |
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| 15:52.02 | Neoti | anyone here use astmanproxy? |
| 15:52.25 | drmessano | Not anymore |
| 15:52.33 | drmessano | It's long extinct in need and use |
| 15:53.03 | ipengineer | I have been but maybe not a requirement anymore? Used to Asterisk would crash with too many connections if astmanproxy wasnt used. |
| 15:53.11 | ipengineer | drmessano: Is this no longer the case? |
| 15:54.07 | drmessano | Asterisk Manager has been beefed up to handle abuse |
| 15:54.28 | drmessano | I'm sure someone can find a use case where it buckles, but I think it's exponentially more difficult now |
| 15:54.36 | ipengineer | Ok good to know! |
| 15:54.37 | Neoti | i have a problem where by astmanproxy the .users file i have entered the to and from context but its not filtering based on this... it just gives me all extensions for the system but i only want extensions on a specific context |
| 15:55.24 | Neoti | well first off if i enter the context it does not allow me to log in... if i leave it blank i can |
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| 16:25.37 | navaismo | Morning All |
| 16:27.30 | drmessano | sup |
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| 16:37.42 | newtonr | howdy |
| 16:38.41 | FuriousGeorge | hey all |
| 16:39.17 | FuriousGeorge | if DTMF sounds play as long as the button is pressed on the phone, then can I safely assume that DTMF duration is not the cause of our DTMF recognition issues? |
| 16:44.19 | Chainsaw | FuriousGeorge: If DTMF detection is intermittent, it's often because it's being passed inband. What phone is this? SIP handset or an analog one? |
| 16:45.36 | FuriousGeorge | Chainsaw: cisco 50Xg series. the sip provider's advice was to pass DTMF inband instead of with the auto setting |
| 16:46.33 | Chainsaw | FuriousGeorge: If you pass SIP out of band, it is a lot easier for Asterisk to follow. Certainly less computationally expensive. |
| 16:46.39 | [TK]D-Fender | You are talking about a phone... and a provider in the same sentence. Each leg is INDEPENDENT |
| 16:47.06 | Chainsaw | FuriousGeorge: But as Fender says, if the SIP provider wants inband, you can do that. Just make sure the phone side is not inband. |
| 16:47.54 | FuriousGeorge | [TK]D-Fender: inhouse calls detect dtmf fine, so the problem must be on the provider's leg |
| 16:48.13 | [TK]D-Fender | FuriousGeorge: So you call in, hit an IVR / voicemail, etc and it doesn't work? |
| 16:48.15 | FuriousGeorge | Chainsaw: so if i understand: asterisk should be set to inband and the phones to rfc2833? |
| 16:48.16 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 16:48.24 | [TK]D-Fender | FuriousGeorge: What codec are your channels using? |
| 16:48.43 | Chainsaw | FuriousGeorge: That seems to be the scenario your SIP provider wants you to try. Personally I wouldn't use inband unless I had to. |
| 16:48.45 | FuriousGeorge | [TK]D-Fender: only on outbound calls is DTMF not detected by 3rd parties. im using ulaw. |
| 16:49.34 | Chainsaw | FuriousGeorge: That's what I had. I have inband DTMF on my ISDN gateways now. |
| 16:49.37 | FuriousGeorge | sip provider is also isp, so they have this trunking device hooked up to it's own modem, and client put all the phones on that LAN, which I cannot access remotely unfortunately, so I can't test now |
| 16:49.42 | Chainsaw | FuriousGeorge: And that fixed it. |
| 16:50.02 | Chainsaw | FuriousGeorge: But don't set the phone side to inband. |
| 16:50.29 | FuriousGeorge | Chainsaw: that's what i'll try as soon as they bridge the LANs and I can get in |
| 16:51.01 | FuriousGeorge | thanks guys |
| 16:57.58 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 17:00.00 | *** join/#asterisk child (b39b4583@gateway/web/freenode/ip.179.155.69.131) |
| 17:01.55 | child | can I multiply the output of a 56k modem as a hub do for internet conection? |
| 17:03.16 | SpeedEvil | child: What do you mean. |
| 17:03.54 | SpeedEvil | If you mean 'can I bond multiple 56K modems to get a faster connection - then yes. |
| 17:05.08 | child | how can I spread lines on the company with just one output ? |
| 17:05.18 | child | is there some hardware? |
| 17:07.14 | SpeedEvil | child: you are not asking an answerable question. What are you trying to do. |
| 17:07.37 | child | I`m sorry I just dont understand enough |
| 17:09.24 | child | I want to create and configure a full central pabx with asterisk to use one voip account for all people at home.... what do I need? |
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| 17:10.57 | child | what I need to configure a number for people dial? |
| 17:11.13 | child | I mean a phisical number |
| 17:11.28 | WIMPy | Numbers aren't physical. |
| 17:11.43 | WIMPy | You need a phon line or an account with an ITSP. |
| 17:11.53 | child | but can you dial by land phone? |
| 17:12.45 | WIMPy | Sure |
| 17:12.54 | child | phone line you mean just the cables or land phone line included included the company? |
| 17:13.09 | WIMPy | Otherwise you don't need anything more than your configuration |
| 17:13.35 | child | and in the way to have 6 phones at home |
| 17:14.00 | WIMPy | You don't know a phone line? |
| 17:14.01 | child | and I have a hardware with 1 line phone |
| 17:14.44 | child | I think I know nothing |
| 17:14.48 | child | hahahah sorry |
| 17:15.17 | *** join/#asterisk pawprint (~pawprint@unaffiliated/pawprint) |
| 17:15.40 | file | wobbles |
| 17:15.41 | rrittgarn | the book might be a good starting point child |
| 17:15.43 | rrittgarn | ~book |
| 17:15.43 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:16.40 | child | to have more than 1 line I need a hardware who offers that? |
| 17:16.57 | WIMPy | Define "line". |
| 17:17.14 | WIMPy | And I don't think that book helps if you don't know the basics. |
| 17:17.28 | child | way to make a call |
| 17:18.16 | child | 2 calls at same times, are done on 2 lines? |
| 17:18.21 | WIMPy | The answer is still the same to make calls you need phone line(s) or account(s) with ITSP(s). |
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| 17:18.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:18.29 | child | or is possible to do with one line at asterisk |
| 17:18.40 | WIMPy | Two POTS line or one BRI. |
| 17:18.46 | child | I mean on account on ITSP |
| 17:19.00 | WIMPy | Asterisk doesn't make a difference as to how many calls you can have. |
| 17:19.07 | WIMPy | Not a positive one at least. |
| 17:19.36 | child | what is pots line |
| 17:19.48 | WIMPy | ~pots |
| 17:19.48 | infobot | [~pots] POTS (Plain Old Telephone Service) is the term for a common analog phone line service as is used world-wide. The "phone company" is called FXO (~fxo), and the user end-point (or phone) is called FXS (~FXS). POTS supports 1 channel, and possibly call-waiting, 3-way calling, CID, as signalled to the telco. |
| 17:19.59 | child | ok |
| 17:20.10 | child | ~bri |
| 17:20.10 | infobot | [~bri] Basic Rate Interface - a form of ISDN that consists of 2 * 64kbit/s Bearer (B) channels and 1 * 16kbit/s signalling channel (D) |
| 17:21.38 | pawprint | anyone have any experience with really old versions of trixbox? i'm running a trixbox version from 2008 (v2.8), and was wondering if i can expect it to come up on its own when rebooted, or if i have to do anything special to get it to run? |
| 17:22.05 | WIMPy | Try #history. |
| 17:22.31 | WIMPy | We don't support GUIs here anyway, but additionally |
| 17:22.43 | WIMPy | ~trixbox |
| 17:22.43 | infobot | Delving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous. Trixbox was one of the earliest complete PBX distros and a relic of a bygone era. While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD. Also, an example of how not to run a business. |
| 17:23.09 | WIMPy | So we can't do anything more than wish you good luck. |
| 17:23.15 | child | and to have 6 calls same time WIMPy ? |
| 17:23.19 | child | 3 bri? |
| 17:23.31 | WIMPy | child: For example. |
| 17:24.09 | child | I need know that to know what hardware need I buy |
| 17:25.09 | WIMPy | Either soem BRIs and a multiport BRI card or one or more ITSP accounts and a decent internet connection. |
| 17:26.50 | child | can I use 6 itsp accounts at same bri, or 3 bri for 1 ITSP accout? |
| 17:27.09 | child | is it? |
| 17:27.12 | WIMPy | It's either one or the other. |
| 17:27.19 | WIMPy | Well, or a mix of them. |
| 17:27.20 | child | yes |
| 17:27.32 | child | wow thats great |
| 17:27.59 | child | is an difficult config to make 6accounts at same bri? |
| 17:28.00 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:28.09 | WIMPy | And what an ITSP offers per account is very different. It might be one or multiple phone numners and it might be one or multiple concurrent calls. |
| 17:28.31 | child | how is possible to use the same line cable and have 6 diferent calls inside it |
| 17:28.34 | child | hahahahahah |
| 17:28.37 | child | ? |
| 17:28.57 | WIMPy | VOIP |
| 17:29.17 | WIMPy | That't he option of using ITSP(s) |
| 17:29.21 | WIMPy | the |
| 17:29.38 | WIMPy | BRIs are phone lines. |
| 17:30.00 | WIMPy | goues out to hunt for some easter eggs. BBS |
| 17:30.28 | child | how about the speed? |
| 17:30.43 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:31.03 | child | depends on hardware? if you think you have 200mb internet conection |
| 17:31.06 | child | ? |
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| 17:50.43 | WIMPy | What speed? |
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| 17:56.38 | child | what the diference between FXE and FXO modules? |
| 17:56.54 | child | FXS |
| 17:57.00 | WIMPy | ~pots |
| 17:57.01 | infobot | [~pots] POTS (Plain Old Telephone Service) is the term for a common analog phone line service as is used world-wide. The "phone company" is called FXO (~fxo), and the user end-point (or phone) is called FXS (~FXS). POTS supports 1 channel, and possibly call-waiting, 3-way calling, CID, as signalled to the telco. |
| 18:00.00 | [TK]D-Fender | child: FXS modules let you plug in PHONES. FXO modules let you plug in LINES |
| 18:00.34 | child | thx [TK]D-Fender |
| 18:01.29 | child | so if I`m thinking to use just voip I don`t need FXO modules? |
| 18:02.04 | [TK]D-Fender | If you don't intend on using an analog line you have... then no, you don't need an FXO interface |
| 18:02.24 | child | nice |
| 18:03.11 | child | and that card for example http://www.ebay.com/itm/Digium-TE122P-Single-Span-T1-E1-J1-PCI-Card-w-Echo-Cancellation/121174724519?_trksid=p2047675.c100009.m1982&_trkparms=aid%3D333002%26algo%3DRIC.CAT%26ao%3D1%26asc%3D20644%26meid%3D5900803693704227500%26pid%3D100009%26prg%3D9209%26rk%3D4%26rkt%3D10%26sd%3D181365216800 |
| 18:03.29 | WIMPy | That's for a PRI. |
| 18:03.34 | WIMPy | ~pri |
| 18:03.34 | infobot | i guess pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 18:04.08 | child | thats not for me so? |
| 18:04.28 | [TK]D-Fender | Do you have a T1/E1/J1 circuit from your telco? |
| 18:04.29 | child | what the max call same time can I do with this hardware |
| 18:04.36 | WIMPy | Probably not. |
| 18:04.43 | WIMPy | 30 |
| 18:04.48 | child | I dont |
| 18:04.48 | [TK]D-Fender | child: depending on how it's configured, 31 |
| 18:05.49 | child | its strange to hear that looking to just one port |
| 18:06.25 | [TK]D-Fender | ~book |
| 18:06.25 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:06.42 | [TK]D-Fender | Go read the book.. it will explain more about basic telephony technologies..... |
| 18:06.56 | [TK]D-Fender | All of this is in there... |
| 18:07.07 | WIMPy | There are different flavours of phone lines. a PRI is the biggest for DIY. |
| 18:07.47 | child | and this hard http://www.ebay.com/itm/TDM410P-with-4FXO-FXS-ports-Asterisk-card-4FXS-400P-Trixbox-Elastix-card-Digium-/320971502245?pt=Phone_Switching_Systems&hash=item4abb6472a5 |
| 18:07.51 | child | is it good? |
| 18:08.26 | WIMPy | POTS is never good. |
| 18:08.55 | WIMPy | Not to connect more than a simple phone or fax machine anyway. |
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| 18:10.27 | [TK]D-Fender | [14:07]childis it good? <- Good for WHAT? |
| 18:10.34 | child | ahahah |
| 18:10.50 | [TK]D-Fender | child: you apparently don't NEED it... so why are you asking? |
| 18:11.09 | child | I do... |
| 18:11.36 | [TK]D-Fender | You just confirmed "only VoIP". That means no "lines" |
| 18:11.43 | WIMPy | For what? |
| 18:11.50 | child | how do you call more than 1 call same time? |
| 18:11.51 | FuriousGeorge | are there any providers that do 100% prorated anymore? X/min/channel inbound and out |
| 18:11.54 | child | more than 1 line? |
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| 18:12.12 | FuriousGeorge | where X ~= 0.01 USD |
| 18:12.17 | [TK]D-Fender | PHYSICAL copper jacks from your LOCAL telephony company mounted in your building <---------- |
| 18:12.18 | child | remember I`m talking about voip only |
| 18:12.28 | [TK]D-Fender | that cards is for TELCO PHONE LINES |
| 18:12.30 | [TK]D-Fender | not voip |
| 18:12.39 | child | ok |
| 18:12.49 | child | can you give me an example for voip? |
| 18:12.55 | [TK]D-Fender | voip = INTERNET |
| 18:12.59 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:13.01 | child | the first was for voip? |
| 18:13.04 | [TK]D-Fender | There is no "magic card" for this |
| 18:13.11 | [TK]D-Fender | child: NO |
| 18:13.40 | [TK]D-Fender | [14:07]childand this hard http://www.ebay.com/itm/TDM410P-with-4FXO-FXS-ports-Asterisk-card-4FXS-400P-Trixbox-Elastix-card-Digium-/320971502245?pt=Phone_Switching_Systems&hash=item4abb6472a5 <- this is for a DIGITAL circuit to your telephone company. NOT VOIP |
| 18:13.56 | [TK]D-Fender | analog, sorry... |
| 18:14.12 | [TK]D-Fender | [14:03]childand that card for example http://www.ebay.com/itm/Digium-TE122P-Single-Span-T1-E1-J1-PCI-Card-w-Echo-Cancellation/121174724519?_trksid=p2047675.c100009.m1982&_trkparms=aid%3D333002%26algo%3DRIC.CAT%26ao%3D1%26asc%3D20644%26meid%3D5900803693704227500%26pid%3D100009%26prg%3D9209%26rk%3D4%26rkt%3D10%26sd%3D181365216800 <- this one was the DIGITAL ONE |
| 18:14.33 | [TK]D-Fender | both are for JACKS you have in your building direct from YOUR local phone company. |
| 18:14.47 | [TK]D-Fender | Different kinds of circuits, but hardware circuits nonetheless |
| 18:19.20 | FuriousGeorge | child: all you need is one of these: http://www.voip-info.org/wiki/view/Sip+Trunking+Providers |
| 18:20.12 | *** join/#asterisk child (b39b4583@gateway/web/freenode/ip.179.155.69.131) |
| 18:20.20 | FuriousGeorge | in order to make and take calls to the public telephone network (e.g. land lines and mobile phones) |
| 18:21.17 | child | ~book |
| 18:21.18 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:21.37 | FuriousGeorge | 99% of SIP (VoIP) providers will allow you to place or receive simultaneous calls, but it's not free |
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| 18:30.35 | Valken | Hi verybody |
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| 21:10.12 | Valken | Hi |
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| 21:14.09 | navaismo | wasaaaaaaa |
| 21:20.12 | navaismo | ¬¬ |
| 21:26.52 | Valken | On asterisk. How can I have an other connected user pick up another phone while it’s ringing (let’s say a secretaty picking up the ringing phone of it’s colleague) ? |
| 21:27.33 | [TK]D-Fender | Valken: "core show application pickup" |
| 21:27.36 | [TK]D-Fender | Valken: "core show application pickupexten" |
| 21:28.07 | Valken | Thank |
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| 21:47.14 | Ta^3 | Anyone here speaks spanish or french? |
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| 21:47.24 | [TK]D-Fender | probably |
| 21:48.00 | Ta^3 | perhaps this could be interesting for you, https://issues.asterisk.org/jira/browse/ASTERISK-23569 |
| 21:48.41 | Ta^3 | it's just trivial stuff: digits/h-billion is not translated properly, it could be wrong in french too, but can't tell. |
| 21:50.16 | navaismo | o/ spanish |
| 21:51.08 | [TK]D-Fender | TaLooks like it has been submitted clearly.... what's thwe question? |
| 21:51.52 | Ta^3 | navaismo, take a look at the bug I just posted. ;-) |
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| 22:20.50 | ipengineer | In mixmonitor is there a way to playback a beep to the callee? |
| 22:24.50 | dym | whilst recording? |
| 22:25.04 | ipengineer | dym: Yes |
| 22:25.13 | dym | as in multiple times? or how? |
| 22:25.46 | ipengineer | Yes as long as the call is recording it is beeping… So if a call comes in and we enable mixmonitor the caller will hear the beeping until the call ends or monitor is disabled |
| 22:25.54 | ipengineer | some legal requirements.. |
| 22:26.30 | dym | You could use Dial's "L" option |
| 22:27.01 | dym | It's used as limiting function, but you could utilise it for your purpose |
| 22:30.05 | navaismo | sounds like this need to be released soon--->https://reviewboard.asterisk.org/r/3362/ |
| 22:35.02 | ipengineer | nanoha-sama: I agree! |
| 22:35.16 | ipengineer | dym: I am curious how that would work for inbound calls though since the DIAL isnt used on those |
| 22:35.50 | dym | Well, you did not provide a complete scenario, did you? :) |
| 22:36.00 | navaismo | ipengineer, in the meantime you can use an script, that script can inject a beep periodically using chanspy |
| 22:36.28 | ipengineer | Ok I will take a look into that. Thanks |
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