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| 11:21.59 | MrBar | where to get good appliance to make call center? |
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| 12:56.56 | ericNL | MrBar, vicidial? |
| 12:57.35 | ericNL | http://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev3 |
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| 13:46.15 | MrBar | ericNL, hi |
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| 14:49.01 | Samot | MrBar: When you say "make call center" what are you actually looking for? |
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| 17:04.43 | kfife | Preferred SIP client for Windows? Anyone? |
| 17:23.23 | MrBar | kfife, none |
| 17:24.02 | kfife | ;-) |
| 17:25.30 | kfife | What about Blink, Yate etc |
| 17:27.11 | kfife | MicroSIP |
| 17:33.14 | Samot | Zoiper, X-Lite, Bria... |
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| 17:35.40 | kfife | Bria and X-Lite kinda suck. |
| 17:36.18 | kfife | X-Lite has ads, and I've got many bria licenses, but it's just too hard to keep proving to them that I've paid for the damned software |
| 17:36.35 | kfife | IMO |
| 17:36.50 | kfife | To download, I've gotta log in to the fucking 'store' |
| 17:36.58 | kfife | and they've deleted my account |
| 17:40.00 | kfife | So now I'ts MY job to prove to them. That would cost me more than what I paid for the license. |
| 17:40.04 | kfife | Fuck that |
| 17:45.03 | *** join/#asterisk mcudev (~mcudev@23-31-69-117-static.hfc.comcastbusiness.net) |
| 17:49.37 | mcudev | Hi all, I am working on a project that requires asterisk to connect two softphones. This is my first experience with asterisk and i have a few questions about the operation |
| 17:49.57 | mcudev | first: is asterisk just a PBX manager, or does it have the ability to run as a softphone? |
| 17:51.00 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 17:51.30 | mcudev | second: I have asterisk 11.7 running on an ubuntu server. I try and start it from the command line in the foreground (-f) and it runs but there are a lot of errors and warnings. Is this expected or do I have to update the config? |
| 17:51.46 | WIMPy | Not sure what a PBX manager is. Maybe somethiong like FreePBX? Asterisk is not. Some call it a PBX, I'd prefer to call it a voice application server or something. And yes, you can use it as phone. |
| 17:52.28 | mcudev | http://pastebin.com/gQ0KeZUF |
| 17:52.35 | WIMPy | Update from what? You write your config. |
| 17:53.05 | mcudev | ok, thanks. after reading some documentation I am going to work on gettting it running as a server and use a different SIP application to confirm it is working |
| 17:53.33 | mcudev | this is a fresh install (apt-get install), so maybe I missed the config |
| 17:54.32 | WIMPy | Don't know what's in that packet, but generelly you shouldn't use any supplied config, but rather see them as examples/documentation. |
| 17:55.36 | mcudev | ok |
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| 18:12.43 | kfife | Ring.cx - ring sip client? Opinions? |
| 18:12.51 | kfife | .cx Christmas Island. Who knew |
| 18:13.18 | kfife | .fz Festivus Island? |
| 18:13.39 | mcudev | so my install seems to be loading modules, like Ldap, how can I disable some of these so I don't get errors at startup? |
| 18:16.12 | mcudev | I assume modules.conf? |
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| 18:23.20 | mpo42vr | Hi guys |
| 18:24.46 | mpo42vr | What is a good practice to get information of calls such as timestamp, call number, duration into an postgres DB? |
| 18:25.05 | kfife | Regular old CDR |
| 18:25.28 | mpo42vr | thanks |
| 18:25.47 | kfife | Just install the module |
| 18:26.56 | [TK]D-Fender | mcudev, modules.conf : noload => module.so |
| 18:27.12 | mcudev | ok, thanks |
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| 18:56.59 | mcudev | alright, I have a functional asterisk server with two users, tested using two SIP softphones. Next step is getting one of the PCs to run asterisk as a softphone |
| 18:57.08 | mcudev | and suggestions on guides to do this? |
| 18:57.26 | mcudev | btw, shitty internet connections means I might drop, but I will reconnect |
| 19:02.06 | [TK]D-Fender | * isn't a softphone |
| 19:06.55 | mcudev | it's server only? |
| 19:14.26 | [TK]D-Fender | It is a Telephony toolkit |
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| 19:15.24 | [TK]D-Fender | * doesn't have a "phone interface" |
| 19:15.45 | [TK]D-Fender | Where's the "hold" button? How would you do DTMF? To tell it to do the things a phone would do? |
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| 19:31.47 | kfife | I adapted a Staples EASY button onto my * serever for hold |
| 19:32.35 | kfife | Plugged an old land line phone into the other big phone jack on the back of the Xeon server. There are 4 phone jacks. |
| 19:33.02 | kfife | Not working. |
| 19:33.06 | kfife | :-) |
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| 19:38.14 | mcudev | I was under the impression that sip connections and audio could be passed using asterisk on one machine to another via an asterisk server |
| 19:38.57 | WIMPy | Yes. There are multiple options. |
| 19:40.19 | mcudev | the reason I am interested in asterisk as the audio sender is I am working on a speaker phone test, and I want to leverage something under active development that can provide some echo cancellation |
| 19:41.01 | kfife | * sounds like a good choice for that. |
| 19:41.41 | [TK]D-Fender | mcudev, Yes you can call from one * to another. That's no different than calling from * to anything else. |
| 19:42.07 | [TK]D-Fender | mcudev, it does NTO give you an interface to actu like a phone. No buttons to push. No integration to the audio interface |
| 19:42.23 | mcudev | that's fine, I am going to do it all from a command line |
| 19:42.34 | [TK]D-Fender | There are already a TON of softphones out there. Why are you thinking you need to turn * into one? |
| 19:42.52 | kfife | and TONS of the softphones are shit. |
| 19:42.54 | mcudev | I read that * has advanced AEC |
| 19:43.02 | mcudev | that's what I was thinking |
| 19:43.14 | [TK]D-Fender | You read wrong. |
| 19:43.17 | mcudev | I am looking at pjsip and linphone |
| 19:43.34 | mcudev | care to correct me? |
| 19:43.40 | [TK]D-Fender | I just did |
| 19:43.47 | [TK]D-Fender | * EX is on ***DAHDI*** |
| 19:43.50 | [TK]D-Fender | not ASTERISK |
| 19:43.56 | [TK]D-Fender | EC |
| 19:43.58 | kfife | Linphone is not suitable to mortals. No dialpad (who needs 'em with SIP uri's right?). The dialpad that CAN be configured to show doesn't work the way users would expect. Also, A,B,C,D DTMF keys anyone? |
| 19:44.03 | kfife | Good for devs though |
| 19:44.10 | [TK]D-Fender | Asterisk does precisely NO EC |
| 19:44.31 | mcudev | I am a C dev |
| 19:44.48 | mcudev | ok |
| 19:45.06 | mcudev | DAHDI = Analog interface for physical phones? |
| 19:45.11 | [TK]D-Fender | yes |
| 19:45.21 | [TK]D-Fender | for TDM interfaces <- |
| 19:45.22 | kfife | MicroSIP is super clean and lightweight. It's unsigned, and has some stupid defaults, like default ringtones & ringback tones that you may want to change (import better sound files, configure etc). Not impossible, but kind of stupid. |
| 19:45.38 | kfife | RING appears to be centered upon thier service, and necessarily brings bonjour. |
| 19:45.47 | mcudev | so if I have a speaker and mic connected to a PC running *somesoftphone* where would I look at doing EC? |
| 19:45.53 | kfife | X-Lite has ads |
| 19:46.07 | kfife | Jitsi client looks promising. |
| 19:46.08 | [TK]D-Fender | mcudev, inside of *somesoftphone* |
| 19:46.15 | [TK]D-Fender | that's why it's an option IN there |
| 19:46.19 | [TK]D-Fender | IT has to do the job |
| 19:46.31 | [TK]D-Fender | You can't ec farter away than the analog interface layer itself |
| 19:46.41 | kfife | Farter. Tee hee... |
| 19:46.42 | [TK]D-Fender | You can't shove some SIP thing in the middle to try to add that |
| 19:46.44 | [TK]D-Fender | it's too late |
| 19:46.50 | [TK]D-Fender | farther* |
| 19:46.59 | kfife | Oh. That makes more sense. |
| 19:47.57 | mcudev | ok |
| 19:48.41 | mcudev | so, since * can be a softphone, there is no EC module/script/thing available, correct? |
| 19:48.54 | [TK]D-Fender | * is not a softphone |
| 19:49.08 | mcudev | omfg |
| 19:49.19 | [TK]D-Fender | NO FUCKING EC <- Am I clear now? |
| 19:49.25 | [TK]D-Fender | NONE |
| 19:49.25 | mcudev | fine |
| 19:49.30 | [TK]D-Fender | GAME OVER |
| 19:49.44 | [TK]D-Fender | Find a better softphone. There are dozens of them |
| 19:52.47 | *** part/#asterisk mcudev (~mcudev@73.95.135.15) |
| 19:55.16 | [TK]D-Fender | "But * can still do all this, right?" |
| 19:57.00 | [TK]D-Fender | People should stop trying to turn a ceramic flower pot into an airplane and just fix another one they have easy access to back into one that flies the way they like |
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| 20:33.18 | Spengler | does anyone in here use asterisk certified? |
| 20:34.59 | [TK]D-Fender | very few and virtually no point. |
| 20:35.08 | [TK]D-Fender | The only point is if you have a service contract with Digium |
| 20:35.17 | [TK]D-Fender | Because barring that there is no advantage |
| 20:51.09 | Spengler | i thought maybe it was better for stability concerns |
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| 20:53.55 | [TK]D-Fender | nope |
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| 22:33.50 | rwb | Hi, Is there a way to set the ringtime to 0 on an extension? Or another way to make the calls to an extension go directly to voicemail without ringing? |
| 22:34.38 | WIMPy | Don't call any device. |
| 22:34.46 | WIMPy | It's your dialplan. |
| 22:37.19 | rwb | I would like to do this programically so I can write a script that will turn extensions "on" or "off" by running a command line script. Any advice? |
| 22:37.49 | rwb | I'm using FreePBX distro. |
| 22:38.02 | WIMPy | Use AstDB. |
| 22:38.12 | WIMPy | Ask in #freepbx. |
| 22:38.22 | rwb | Ahh, Thanks! |
| 22:52.08 | rwb | In general what is the best approach to having an extension go directly to another destination? |
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| 23:59.13 | adonaros | Good day |