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| 03:59.44 | backlander | hello folks, i'm trying to send voicemail message to my email address, unfortunately my isp is blocking all outbound traffic on port 25. What is the easiest solution? |
| 04:02.52 | backlander | continues his search engine discovery quest ;-) |
| 04:03.08 | Quadrant | probably smtp + ssl |
| 04:03.25 | Quadrant | or set up smtp relay to your isp's smtp server |
| 04:04.02 | backlander | can that be done with asterisk alone? |
| 04:04.11 | Quadrant | no you need to do it in your MTA |
| 04:04.23 | Quadrant | sendmail, postfix, exim, etc. |
| 04:05.06 | backlander | ok thanks for the information Quadrant |
| 04:08.03 | Dovid | anyone here use voipmonitor ? |
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| 04:10.58 | smeet2002 | hi people |
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| 04:32.13 | ChannelZ | hi person |
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| 04:44.56 | smeet2002 | DOes anybody use siptosis? |
| 04:47.12 | smeet2002 | OK..I am going to sleep as well |
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| 05:24.55 | esaym153 | where are passwords for voicemails stored? |
| 05:25.05 | esaym153 | err wait, it is in the voicemail.conf right? |
| 05:26.10 | carrar | we can't tell you |
| 05:26.14 | carrar | it's secret!! |
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| 05:37.07 | esaym153 | :( |
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| 06:13.29 | atan | esaym153, voicemail.conf lol |
| 06:14.39 | esaym153 | yea I noticed, but I also noticed that the voice menu in a user's voice mail box gives an option to change the password. So I am guessing that it would then be stored in the asterisk db maybe? |
| 06:16.36 | atan | No. That is stored in voicemail.conf as well. |
| 06:16.51 | atan | Where is voicemail.conf, /etc/asterisk/ ? |
| 06:17.04 | atan | cd /etc/asterisk/ && ls -la | grep voicemail.conf |
| 06:17.24 | atan | Then ps aux | grep asterisk |
| 06:17.36 | atan | It won't change anything, but post your output |
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| 06:18.26 | raden | YOOOOOOOo |
| 06:18.37 | atan | esaym153, long story short I'm getting at your permissions. |
| 06:18.53 | atan | Asterisk needs to be able to write to the voicemail.conf file. If it's now owned by Asterisk, it can't write to it. |
| 06:19.07 | atan | However it will hold it in memory until it is able to, so the user can still login with their new password. |
| 06:19.21 | atan | You are best off correcting file permissions though. |
| 06:19.29 | atan | raden, wassssssssssssssssup |
| 06:19.42 | atan | s/now/not/ |
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| 06:20.42 | raden | same ole |
| 06:21.00 | esaym153 | atan: oh, interesting |
| 06:21.22 | atan | esaym153, I take it things work as expected now? :P |
| 06:21.55 | atan | raden, you haven't an idea how I can play DivX movies to video phones when they dial in? eg. Dial 123 for the movie line, then select from the IVR which movie they want? lol |
| 06:22.02 | esaym153 | how is asterisk 1.6? I just upgrading from 1.4 and I am having some issues. Mainly music on hold dies for no reason and some ivr's also die unexpectatly. verbose and debug output shows nothing... |
| 06:22.18 | esaym153 | atan: yea, I wasn't worried about it, just curious |
| 06:22.21 | atan | esaym153, I'm on 1.8.1. ^_^ |
| 06:22.41 | atan | esaym153, not that that means anything... but I've found a few useful new additions in 1.8.1 I swear I can't live without! |
| 06:22.50 | esaym153 | oh heck, I was thinking 1.6 was 1.8... dang it... |
| 06:23.12 | atan | <3 #1.8.1 with all his <3 |
| 06:23.27 | atan | Err, s/#/*/ even |
| 06:23.38 | esaym153 | how long is 1.4 supported to? |
| 06:23.55 | raden | atan, i would think there would be a function for it |
| 06:24.02 | raden | where Katty ? |
| 06:25.29 | atan | esaym153, can't recall. I saw a nice table somewhere about it. I think it was like 2012 or something but I'm likely wrong. |
| 06:25.45 | atan | raden, does Mrs. Katty about movies over SIP? ^_^ |
| 06:26.04 | raden | I dont know she might |
| 06:26.15 | raden | misses Katty |
| 06:26.30 | atan | Errrr.... I turn on the "blonde" radio station thinking I'll hear some Pink and I get Beiber. Just my luck. |
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| 07:38.31 | shnootz | hi i have Xorcom PBX which i installed PIAF on it and i cannot make the astribanks work |
| 07:39.02 | shnootz | dahdi_hardware -v shows the devices as PRI unknown |
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| 07:39.21 | shnootz | can anyone help me? |
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| 08:47.23 | atan | If you couldn't get local numbers through a VOIP provider, is there any way you could buy from local cellular companies + purchase their 'unlimited inbound' service for like $10 or something, and stick the GSM card into some decive that can register as a SIP device? |
| 08:47.48 | atan | Obviously it might not make sense to route your outbound calls over it (unless maybe all your trunks are down.. hmm!!) but to get inbound calls at least. |
| 08:49.17 | lauris | yes, that is possible |
| 08:49.30 | lauris | such devices costs ~300gbp |
| 08:54.09 | atan | lauris, you're kidding me? |
| 08:54.13 | atan | What is such a device named? |
| 08:54.20 | lauris | no, i'm not |
| 08:54.57 | lauris | for example this one: http://www.2n.cz/en/products/gsm-gateways/voip/voiceblue-next/ |
| 08:55.30 | atan | The SMS thing would be super cool. Hmm! |
| 08:56.14 | atan | What about something like http://www.dealextreme.com/details.dx/sku.9674 ? |
| 08:56.17 | lauris | it has support for SMS as well |
| 08:56.18 | atan | Only $185 |
| 08:56.52 | lauris | see photos - it's USB ! |
| 08:57.07 | lauris | and the only supported operating system is Windows |
| 08:57.22 | lauris | if you need cheap solution you can build it on your own |
| 08:57.26 | atan | And this VoiceBlue isn't? How does it connect to the PBX? =) |
| 08:57.26 | lauris | with some cheap cellphone |
| 08:58.19 | lauris | atan, voiceblue works like IP device |
| 08:58.47 | lauris | just like any ATA device :) |
| 08:59.22 | atan | I wonder if a WRT54G3G would work O_O |
| 08:59.42 | lauris | have no idea |
| 08:59.47 | lauris | don't think so. |
| 09:00.49 | lauris | atan, if your budget is that small there are no solutions :) |
| 09:01.04 | lauris | no *good working* solutions |
| 09:01.11 | atan | lauris, well for one connection $300 might be fine. But I'm thinking on a larger scale. |
| 09:01.24 | lauris | you didn't mentioned that |
| 09:01.26 | atan | See in my area we can't get local numbers =( but the cell providers have a ton of them |
| 09:01.50 | atan | The issues is that I would _like_ to get local DIDs somehow. If I need to rob them off a GSM provider... fine! lol |
| 09:02.07 | lauris | how many channels do you need ? |
| 09:02.24 | lauris | since when GSM numbers are "local DIDs" ? |
| 09:02.26 | atan | To start out I only need one or two inbound. I believe they only put one phone number on each sim right? |
| 09:02.39 | atan | lauris, maybe I'm thinking about this all wrong. |
| 09:03.03 | atan | lauris, the telco providers will sell you a SIM card for your phone with unlimited inbound for like $20/month. Seems like a steal, really. |
| 09:03.15 | lauris | the same company has a hardware for large scale solutions |
| 09:03.17 | atan | I want to convert that cellphone sim card into a SIP connection so I can use it within Asterisk. |
| 09:03.33 | lauris | http://www.2n.cz/en/products/gsm-gateways/voip/stargate-voip/ |
| 09:04.04 | atan | That 2N® VoiceBlue Next only supports 1 sim card right? So "1 number" per se? |
| 09:04.11 | lauris | see specs |
| 09:04.19 | lauris | as far as i remember - yes |
| 09:04.28 | atan | No. of voice channels 2. Does this mean it takes two sim cards? |
| 09:05.42 | lauris | yes |
| 09:06.11 | lauris | but if two cards are used, both of them must have an identical pin |
| 09:06.20 | lauris | or pin request must be disabled |
| 09:06.32 | atan | Woah. I do not understand. What is this PIN thing? |
| 09:06.46 | atan | The code used to unlock the SIM card? |
| 09:07.00 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 09:07.20 | lauris | yes |
| 09:07.35 | atan | You can disable that on most cards these days can't you? =) |
| 09:07.46 | lauris | if you don't understand what PIN stands for probably this solution is not suited for you :) |
| 09:07.52 | lauris | yes, you can |
| 09:10.02 | atan | Their site lets you add to cart but does not show pricing. Hmm. |
| 09:11.09 | lauris | search for their partners and ask for quote |
| 09:12.34 | atan | http://cgi.ebay.com/Improved-Fixed-Cellular-Terminal-Gateway-GSM-Dual-Band-/260707607421?pt=AU_Home_Personal_Security&hash=item3cb362637d $45? O_O |
| 09:12.59 | lauris | so what ? |
| 09:13.33 | lauris | this one looks like a dumb FXS gateway with no SIP support |
| 09:15.35 | atan | Hmm. My bad. Anyway, thanks for the details! I can't wait to try this out :D |
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| 10:05.51 | GhOnDiE | how can i detect when a call has been picked up using the ami? |
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| 11:00.14 | GhOnDiE | has anybody got any ideas how i can have a light come on if calls are stacked up |
| 11:00.20 | GhOnDiE | ? |
| 11:00.40 | GhOnDiE | dont have to be a light but something that i can control in hardware |
| 11:01.05 | GhOnDiE | or something that sets a pin on a paralell port to high for instance |
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| 11:05.27 | atan | GhOnDiE, that sounds really cool actually. =D |
| 11:12.56 | GhOnDiE | well |
| 11:13.04 | GhOnDiE | im a tech with a radio station |
| 11:13.19 | GhOnDiE | we have a screen that indicates mic live, the time weather etc |
| 11:13.39 | GhOnDiE | that is signalled by a hardware box that i made |
| 11:13.45 | tzafrir | GhOnDiE, did you look at the events you get? |
| 11:13.52 | GhOnDiE | just some basic relay signalling into a joystick port |
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| 11:17.51 | GhOnDiE | ive only looked at the detail of it on voip-info so far but its pretty confusing, im just going to look at a live call now i now how to login etc |
| 11:18.53 | GhOnDiE | i would just plug the hardware into the fxs port but it dont put out enough voltage to switch the relay, my other option will be to rebuild the box and use an external power supply and some transistor switching |
| 11:21.47 | atan | GhOnDiE, not sure if it's worth it but I think you can adjust the voicemail indicator light on most phones |
| 11:22.03 | atan | You could set that line to a custom mailbox that throws a fake file in there to turn the little led on. |
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| 11:35.24 | drmessano | GhOnDiE, are you just trying to get a ring indication or are there some specific call limits you're wanting to trigger the overhead board? |
| 11:36.03 | GhOnDiE | well basically i just want it to flash in time with a specific extension |
| 11:36.32 | GhOnDiE | even if that extension is on the phone |
| 11:36.39 | GhOnDiE | kinda like a call waiting light |
| 11:38.01 | GhOnDiE | so call comes in asterisk answers call > light flashes and phone starts ringing > studio answers phone and light goes off. while one the call another call comes in and light will then flash until they answer it or untill all calls have been answered |
| 11:38.24 | GhOnDiE | i have thought maybe using an fxs port tied into a ring group |
| 11:38.29 | drmessano | All we've ever done is buy Radio Shack Fone Flasher modules, gutted them, put them on the 2 request lines, and fed them to the same relay.. if either request line had a call, the light would flash |
| 11:38.55 | GhOnDiE | however the light needs to flash when an internal call comes in too |
| 11:39.18 | GhOnDiE | was that into an fxs port or direct into the main incoming line? |
| 11:39.38 | GhOnDiE | we have 1 request line that fails over on busy to a sip line |
| 11:40.43 | drmessano | We've used analog lines, so right on the incoming line |
| 11:41.36 | drmessano | But of course, you could set up an FXS port to ring WITH the request line extension, and have that feed a 99 cent relay |
| 11:43.07 | GhOnDiE | ok that was pretty much plan b |
| 11:43.37 | GhOnDiE | i was hoping to do it without having to modify the bow that im already using |
| 11:43.46 | GhOnDiE | box* |
| 11:43.53 | GhOnDiE | thanks for your ideas and help |
| 11:45.42 | GhOnDiE | hmm the astribank does i/o |
| 11:45.45 | GhOnDiE | interesting |
| 11:45.50 | GhOnDiE | but a bit much for my needs |
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| 12:50.33 | WIMPy | Has the issue tracker just disappeared? |
| 13:04.35 | *** join/#asterisk glam (~glam@116.24.119.214) |
| 13:13.24 | atan | https://issues.asterisk.org/main_page.php is up for me WIMPy |
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| 13:14.06 | GhOnDiE | indeed same here too |
| 13:15.24 | atan | GhOnDiE, were you here when they were talking about blocking WIMPy's IP due to too many bug reports? |
| 13:15.33 | GhOnDiE | nope |
| 13:15.39 | GhOnDiE | lol |
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| 15:07.39 | atan | Does the AGI support speaking words outside of the .gsm files it already has? |
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| 15:25.07 | delx | Hey |
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| 15:29.02 | delx | I have a problem with my extensions: Call from 'companyXY' to extension 's' rejected because extension not found. My extensions.conf line looks like this: exten => _X.,1,AGI(/etc/asterisk/forward_call.php) <- Isn't _X matching a string like "companyXY" ? |
| 15:31.16 | delx | ah, working now with the 's' extension! |
| 15:31.19 | delx | thx anyway. |
| 15:32.15 | Chilibeta | Hello there, I am new to asterisk and have an interest in how visual voicemail can be realized with asterisk. |
| 15:32.36 | Chilibeta | Is this the right place to ask here or would be the forums/maling-list a better place for that. |
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| 15:56.45 | atan | I don't support anyone knows which format files should be saved in using Audacity? |
| 15:57.02 | WIMPy | The one you chose. |
| 15:57.09 | atan | Great. |
| 15:57.24 | atan | Any idea which one the default .gsm files are stored in? |
| 15:57.35 | atan | I see several GSM options within audacity |
| 15:58.57 | atan | I get format_wav.c:124 check_header: Not a wav file 49 when I try to open it after saving as GSM 6.1 |
| 15:59.30 | WIMPy | Haven't tried writing gsm. |
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| 16:28.53 | *** join/#asterisk ghostnik11 (~king@pool-173-68-138-36.nycmny.fios.verizon.net) |
| 16:31.16 | ghostnik11 | hi i am trying to replace my home land line the thing is i know i need a piece of hardware that can connect to my router then I can plug my home phone in and I will be able to make free calls through internet but i don't know the proper name of the hardware |
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| 16:32.15 | WIMPy | An ATA for example. |
| 16:34.13 | ghostnik11 | wimpy: exactly is that the name, b/c i want to buy one |
| 16:35.13 | WIMPy | There are many flavours to choose from. You can also get them as PCI cards. |
| 16:36.25 | ghostnik11 | wimpy: okay well is there a flavour that i can use an ethernet cord from router to an ata then have all the phones in the house connect to the ata |
| 16:37.07 | WIMPy | That's the ATAs. |
| 16:37.35 | WIMPy | Take a look aroud some of the voip stores. They probably have tons to choose from. |
| 16:39.12 | ghostnik11 | wimpy: well i mean are ATAs stand alone or will i have to connect it to a computer, b/c i want to just connect it to the router and thats it, simlar to a magic jack only thing i don't what to connect the magic to my computer i want to connect it to a router but my router doesn't have a usb port |
| 16:39.36 | WIMPy | They are standalone. |
| 16:39.45 | ghostnik11 | wimpy: thanks |
| 16:40.30 | WIMPy | They only need an IP connection to some SIP account. |
| 16:40.42 | WIMPy | Well, it need not be sip, but usually will be. |
| 16:42.58 | ghostnik11 | wimpy: well i have two sip accounts one with ekiga and the other with iptel but the thing is my mom wants to reduce her phone bill i told her i might be able to make it possible for her so now i am researching how i will accomplish this feat as i think i can just have my sip account which is linked to google voice place the calls outgoing and in turn have google voice route the call to my sip address |
| 16:44.20 | WIMPy | No idea about the google stuff, but if you have SIP accounts that's already the big portion of the deal done. |
| 16:45.02 | ghostnik11 | wimpy: thanks |
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| 16:45.56 | atan | ghostnik11, careful with the gvoice stuff I hear they've banned a few people from connecting |
| 16:46.10 | ghostnik11 | atan: what do you mean? |
| 16:46.10 | atan | ghostnik11, but fun to setup for personal use nevertheless! :P |
| 16:46.29 | atan | ghostnik11, I hear people were connecting up via jabber to make voice calls, but they blocked their ips after awhile. |
| 16:47.07 | atan | ghostnik11, you might also want an adapter with FXO+FXS if you plan for her to use her current lines |
| 16:47.42 | ghostnik11 | atan: wait basically they found a way to figure out how to make free calls, and google blocked them or there phone companies |
| 16:47.50 | atan | WIMPy, any idea inside AGI how many times a while() loop runs? Like, err, if I'm sitting in a while() loop waiting for user input I want to know how many times the loop runs. If it's once per second, great, 50 timse per second... I might not want to query a db each time :P |
| 16:48.07 | atan | ghostnik11, I do believe Google would have been the one causing the blockage |
| 16:48.24 | atan | ghostnik11, just don't route tons of calls over it or expect your account to get shut down :P lol =) |
| 16:48.32 | ghostnik11 | atan: i thought google was suppose to be transparent |
| 16:49.14 | WIMPy | atan: Just count them. |
| 16:49.40 | atan | "transparent" ? They've offered a trunk for calls... it costs them money, and they aren't going to let the world rape them for free phone service ;) |
| 16:49.45 | ghostnik11 | atan: well i make free calls through google voice for both outgoing and incoming but i don't make 1000's of calls like i am a company just for personal use, i don't even think i use 200 mins a month b/c i usually text a lot through google voice and google talk |
| 16:50.10 | atan | ghostnik11, then I suspect you will be just fine =) |
| 16:51.36 | ghostnik11 | atan: but that might be a problem for my mom, who uses the home phone quite a bit, but i don't think she goes over a 1000 mins a month and if i use pbxes.org they allow up to 2000 per month for free users |
| 16:53.55 | ghostnik11 | atan: oh and about google trunk that they let people talk on for free i think some how google is getting paid by someone that public doesn't know are is in fine print b/c they basically let everyone in the usa and canada i think make free calls and for more advanced people around the world who know how to get around google also them that can make free calls through google voice on the low without google knowing |
| 16:54.24 | atan | The hell. What is this 2000 minutes free thing. 2000 minutes to where? |
| 16:55.06 | atan | Err. My bad. It's not for calls it's just the PBX. |
| 16:55.13 | atan | goes back into his hole in the ground |
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| 16:58.35 | ghostnik11 | atan: yeah pbxes i have hooked up with my sipdroid client on my cell phone with just a data plan offers 2000 minutes a month for free users more for paid users |
| 17:01.21 | ghostnik11 | atan; yeah go to pbxes.com or pbxes.org doesn't matter which one and look them up |
| 17:02.56 | atan | Okay I need a prompt that says "now enter the expiry month" and "now enter the expiry year" but none are included in the sound pack. They have cc-visa, cc-mastercard, cc-discover... I'd think they would have these as well! =D |
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| 17:14.53 | ghostnik11 | wimpy: you said voip stores, do you know any that are in usa |
| 17:16.04 | WIMPy | Sorry, I'm rest of the world. |
| 17:17.45 | ghostnik11 | wimpy: well the rest of the world has it good b/c in the states i think the phone companies don't want people to figure out that they could be making free calls instead of paying them, so all the good voip stuff always comes out in the rest of the world while people like me in the states are just catching on and trying to get the parts |
| 17:18.47 | WIMPy | I think there are a lot more reasons :-) |
| 17:19.03 | ghostnik11 | wimpy: what are those reasons? |
| 17:19.36 | WIMPy | Here they need not fear. 1. it's really hard to get any connectivity without free natinal calls included and 2. tehy use voip themselves on the last mile. |
| 17:20.12 | WIMPy | An antiquated phone system. |
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| 17:22.35 | ghostnik11 | wimpy: oh, so the phone companies in the rest of the world use voip, then that means its cheaper for those people in those countries also its already set up for those people while over here in the states the phone companies like the antique phone system b/c there customers are all tied in, my family has been using verizon since it was bell or nynex or what ever it was b/4 its big merger into verizon |
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| 17:23.27 | ghostnik11 | wimpy: thats over 19 years of a customer and it keeps counting, b/c no one else realizes that they are other options |
| 17:23.35 | WIMPy | The situation is really extremely diffrent. |
| 17:24.40 | WIMPy | In fact I'd say there aren't too many similarities. But I got used to that being in here :-) |
| 17:27.13 | ghostnik11 | wimpy: b/c you guys in here are educated and were willing to figure out your options over here in the states its like we all just go along with these big phone companies b/c if we switch we switch to another big phone company and don't know about voip but if we all in the states started doing voip then the big phone companies would have to drop there rates and plan and then we could all save money in this damn horrible econo |
| 17:27.14 | ghostnik11 | my |
| 17:28.21 | WIMPy | Well, most ppl here don;t know about voip, either. They use it without knowing. |
| 17:29.28 | ghostnik11 | wimpy: i don't get it, there in asterisk chat channel who is leading the charge in voip and don't know about it but are using asterisk servers to make free phone calling |
| 17:29.41 | p3nguin | Like Magic Jack and Vonage. They don't know it's VoIP... they just know it as being Magic Jack or Vonage. |
| 17:30.37 | ghostnik11 | p3nguin: but why wouldn't they go and look up what they buy b/4 they get it and see its voip and that its an option in terms of phone service vs. the big phone companies |
| 17:31.03 | WIMPy | ghostnik11: I talking about customers here. Not ppl in th channel. |
| 17:31.24 | ghostnik11 | wimpy: oh, my fault |
| 17:43.35 | atan | I have a friend who swears by his MagicJack. I cry every time I see him plug it in to his computer to make a call. =( |
| 17:44.42 | Kobaz | you know magicjack has (or had) in their eula that they can spy on your calls, or something like that |
| 17:45.56 | Kobaz | swilson06: you can't do some random name in c, what are you referring to? |
| 17:45.57 | Kobaz | er |
| 17:46.45 | doolph | lol |
| 17:46.50 | doolph | magicjack |
| 17:46.52 | doolph | that is pretty old |
| 17:46.54 | doolph | thing |
| 17:47.13 | atan | doolph, they keep them on locked pegs at the local stores. As if someone would steal it, really. |
| 17:47.35 | doolph | rofl |
| 17:47.43 | atan | Kobaz, I thought it was common for any service provider to spy on your calls so long as they do not repeat what they hear and it's for quality testing reasons. |
| 17:47.47 | WIMPy | always gets a crisis when I see USB headphones advertised as VOIP phones. |
| 17:48.10 | WIMPy | The consumer watchdogs should really do something about such shit. |
| 17:48.31 | doolph | anyone here has a atcom ip0x? |
| 17:48.37 | atan | Yeah, seriously. There are like 500 Linksys Wireless "VOIP" phones for sale at a local repo depo except they _only_ work via USB if you have Yahoo messenger. |
| 17:48.45 | atan | I was wanting to setup some SIP firmware on them so bad. |
| 17:48.51 | atan | They are like $2/ea |
| 17:58.23 | Kobaz | atan: depends on the state |
| 17:59.03 | atan | Kobaz, well I best stop snooping on everyone! lol. Actually it's such a waste of time. Even if they know they are being QAd they are still no fun. |
| 17:59.53 | atan | I think the best thing was hearing one girl at the call center talk about leaving her boyfriend with someone while she should have been working, and he was sitting like just a few rows away where he could not hear her |
| 17:59.55 | atan | And he did not know. |
| 17:59.59 | atan | Man, that was wild. |
| 18:00.14 | Kobaz | heh |
| 18:00.36 | atan | She was reminded that work phones are for work of course :P :D |
| 18:07.57 | Kobaz | a girl got fired from the last place i worked at for being on the phone and crying all day |
| 18:09.15 | atan | I think that's the time you break out the tt-monkeys |
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| 18:12.48 | *** join/#asterisk acxty (be06c31c@gateway/web/freenode/ip.190.6.195.28) |
| 18:13.17 | acxty | Hi guys, does someone know of a open source web phone? |
| 18:15.15 | atan | acxty, Ekiga |
| 18:15.26 | atan | http://ekiga.org/ |
| 18:17.05 | acxty | atan: I am reading that is a softphone, what I am looking is for a webphone |
| 18:17.59 | atan | The hell is the difference? |
| 18:18.54 | acxty | atan: I need it to run on a web page |
| 18:19.15 | atan | Ohhhhh. Interesting. =) |
| 18:20.44 | atan | http://tringme.com/ |
| 18:21.08 | atan | Anyway. I'm not sure but you have me wondering about this now. Neat idea =) |
| 18:21.11 | atan | Can I ask what you' |
| 18:21.15 | atan | 're going to use it for? |
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| 18:22.12 | acxty | I want it to connect to asterisk and create diferent sip accounts for some clients |
| 18:22.24 | acxty | I know it can be done using any softphone |
| 18:22.39 | acxty | but the idea is to use a web application for all the stuff |
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| 18:31.06 | mr-m | anyone know how to get sip traces off a mitel icp3300 without setting up a port-mirror? |
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| 19:11.26 | pabelanger | mr-m: sip ssp set level 3 |
| 19:11.32 | pabelanger | sip ssp set storage c |
| 19:11.39 | pabelanger | sip ssp trace on |
| 19:11.56 | pabelanger | remember to turn it off when done |
| 19:11.57 | pabelanger | sip ssp trace off |
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| 19:39.12 | WIMPy | Wow. 150 EUR for a TE110P today. |
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| 19:51.37 | esaym153 | what would be the proper way to create a user for the asterisk daemon? What would the home directory,password,and shell be? |
| 19:52.15 | WIMPy | "Asterisk as user" might be a good term to throw at google. |
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| 19:52.46 | esaym153 | but but but...Isn't it Christmas?? |
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| 19:53.15 | WIMPy | NFI, but tuesday night is jul. |
| 19:54.28 | esaym153 | I was really just asking more of a general question I guess. more like "how do I create users for daemons" |
| 19:54.36 | Chilibeta | Hello there, someone open to give me some asterisk insights about voicemail/recording interface and such? Anwsering a few questions? |
| 19:55.11 | WIMPy | esaym153: As with all things on *nix it's a question of personal taste. |
| 19:55.19 | WIMPy | ~ask |
| 19:55.20 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:55.24 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 19:55.27 | WIMPy | Chilibeta: ^^ |
| 19:56.24 | Chilibeta | Allright. I am investigating if a (visual) voicemail extension is feasible for a one person project, timespan 6 months. |
| 19:56.42 | WIMPy | What's that supposed to be? |
| 19:57.10 | Chilibeta | Well. Afaik asterisk doesnt support that right now. Perhaps you have heard about the iphone visual voicemail thingy. |
| 19:57.25 | WIMPy | nope |
| 19:57.28 | Chilibeta | Browsing, editing, answering machine/voicemail messages. |
| 19:58.02 | esaym153 | WIMPy: well I just cheated and looked at how the asterisk.deb does it on debian... |
| 19:58.12 | WIMPy | I think I have heard about web based VM GUIs. |
| 19:58.13 | Chilibeta | Editing means deleting, all in a visual fassion from a smartphone client. |
| 19:59.34 | WIMPy | has thought baout doing something for his phones minibrowser, but didn't bother so far. |
| 19:59.34 | Chilibeta | What I have found is a recommendation ( not a real standard ) by omtp and they use a modified imap protokoll to access voicemail content. |
| 20:00.07 | WIMPy | You can use imap storage for Asterisk Voicemail. |
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| 20:01.05 | Chilibeta | Well, perhaps you can outline that a bit more. Asterisk already has some way to access imap? |
| 20:01.29 | WIMPy | Sorry, I never tried that, but it is possible. |
| 20:01.51 | Chilibeta | Allright, great info to dig into and see what is already possible. |
| 20:02.41 | Chilibeta | How I understand my project/topic so far is, the other way arround. Not asterisk storing voicemails in imap, rather adding |
| 20:03.39 | WIMPy | Whtever it is, I'd guess it should be pretty straightforward. |
| 20:03.44 | Chilibeta | an imap layer (as plugin/whatever) to asterisk, so that it is possible to access all the voicemail storage via imap. |
| 20:04.46 | WIMPy | It's more of an storage backend for Voicemail. |
| 20:05.22 | WIMPy | But I haven't taken a look at what it is. |
| 20:06.04 | Chilibeta | Hmm... do you know under which name this extensions is provided or is imap the preferred storage method for, e.g. voicemail |
| 20:06.29 | WIMPy | It's one of the storage backend options. |
| 20:06.50 | Chilibeta | I see. I will have a look into the docs to see what is there. |
| 20:07.12 | timholum1 | Chilibeta: one other thing you might be able to do is set up an external email account ( gmail , zimbra ext... ) and then just put everyone's email in to that account ( in voicemail.conf ) then you would have an imap storage of all voicemail's |
| 20:07.34 | timholum1 | I just got in the room thought so this might have already been discused, if so sorry |
| 20:08.33 | Chilibeta | No, its fine. You just mentioned what we are thinking of that this exists. |
| 20:09.20 | WIMPy | That's not the same thing. |
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| 20:12.07 | Chilibeta | I am thinking of something more specialized for voicemails accessed via a nice smartphone app. |
| 20:12.54 | Chilibeta | So that it would be possible to browse/delete/listen them. |
| 20:13.02 | timholum1 | you could use a mysql storage backend and just use a web interface |
| 20:14.02 | timholum1 | it would involve some development ( proabably php,python or perl knolage as well ) |
| 20:14.14 | Chilibeta | Well, yes. |
| 20:14.46 | Chilibeta | But the whole idea is, as the omtp recommendation paper for visual voicemail is backed by numerous telcos. |
| 20:15.24 | Chilibeta | So it could be possible, if a client impl is there, this impl can access more than one voicemail service. |
| 20:16.11 | WIMPy | If you;ve got the specs that shouldn't be hard to do. |
| 20:16.56 | Chilibeta | Yes, thats what I am thinking. Mostly a matter of how much effort. |
| 20:18.03 | Chilibeta | What there is in the android market, is an app to access asteriskv(oice)m(ail) via a recording interface. It seems a bit basic from the screen shots, but its there. |
| 20:18.40 | Chilibeta | I have searched quickly the asterisk docs, but havent found something like a recording interface. |
| 20:18.50 | Chilibeta | Does someone know what that could be? |
| 20:19.05 | WIMPy | Probably because there's no need. |
| 20:19.27 | WIMPy | You would be accessing the backends directly, I guess. |
| 20:19.52 | timholum1 | or maybe use ami? |
| 20:20.06 | Chilibeta | What is ami? Never heard of that. |
| 20:20.10 | WIMPy | Does it have anything about VM? |
| 20:20.18 | WIMPy | The Asterisk Management Interface. |
| 20:21.25 | WIMPy | The only possible catch I see is that you might sabotage the event driven MWI. But I think you can still configure it to the old way. |
| 20:21.35 | timholum1 | it basicly is an interface that you can do just about anything in asterisk through, originate calls, transfer calls ext...... |
| 20:21.55 | timholum1 | one sec let me check |
| 20:22.31 | timholum1 | http://www.the-asterisk-book.com/unstable/asterisk-manager-api.html |
| 20:22.48 | timholum1 | it at least shows how to check the number of voicemails |
| 20:23.22 | timholum1 | if I where you I would enable ajam on your server ( essencialy a web interface to ami ) and play arround in there |
| 20:23.29 | WIMPy | Yes, but I don't see the neccessity to ask. |
| 20:24.18 | Chilibeta | Allright, I see that. Nice, this AMI. |
| 20:25.15 | timholum1 | well I am going to go and play some call of duty, talk to you guys latter |
| 20:26.47 | Chilibeta | Perhaps its a good idea to look through the asterisk-book and to see what is already there. |
| 20:27.06 | WIMPy | That sounds like a plan. |
| 20:27.44 | Chilibeta | And afterwards deciding on the if and how. |
| 20:29.52 | Chilibeta | Thanks a lot you guys for the given insights. Here is night and I will go to bed. |
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| 20:43.08 | xSmurf | is there a way to detect a failure with VMAuthenticate?? |
| 20:43.34 | xSmurf | It doesn't seem to have a jump option like Authenticate |
| 20:48.49 | p3nguin | You could use Authenticate() prior to your vm stuff without authentication. |
| 20:53.10 | xSmurf | huh? |
| 20:53.37 | xSmurf | I use VMAuth and I want to send the user somewhere when it fails |
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| 20:58.51 | esaym153 | the process that writes monitor files I assume rans are the same user as the asterisk process right? Because I am getting a permision denied error in the logs when the monitor process tries to write the file to the directory even though it has write permissions... |
| 21:03.24 | esaym153 | hmm, the directory is symlinked. Can it not follow it or something? |
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| 21:05.07 | p3nguin | You'll have to have the correct permissions on parent directories, as well. |
| 21:06.41 | esaym153 | hmm, that could be it, let me see |
| 21:09.00 | esaym153 | yea that is probably it, |
| 21:09.32 | p3nguin | namei can help reveal all the perms in the path. |
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| 22:12.37 | vasanthv161 | I am an asterisk newbie. I've installed asterisk on ubuntu and now I am looking to try out some basic stuff. Any pointers on where to get started? I've read some documentation so far. But still not quite clear on where to start. |
| 22:12.45 | vasanthv161 | https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality is something I am looking at right now |
| 22:18.16 | vasanthv161 | anyone ? |
| 22:20.23 | tzafrir | vasanthv161, what version (asterisk, ubuntu)? |
| 22:21.07 | tzafrir | what phone(s) do you have? Any hardware phone? Or just the sound card and a SIP client? |
| 22:21.42 | vasanthv161 | ubuntu 10.10 server, Asterisk 1.6.2.7-1ubuntu1 |
| 22:22.26 | vasanthv161 | I am not quite sure what my goal is, but something like connecting google talk / skype with asterisk. |
| 22:23.40 | vasanthv161 | I dont have any hardware phone yet. What is a SIP client? Is that a separate software? Or is Google Talk application a SIP client? |
| 22:23.48 | tzafrir | skype would generally require jumping over some hoops and paying some money |
| 22:24.03 | tzafrir | google talk: did you check the documentation? |
| 22:24.07 | vasanthv161 | tzafrir: I can come to skype later. Maybe google talk first. |
| 22:24.40 | tzafrir | A SIP client is a software phone. On Ubuntu this is mainly Ekiga and Twinkle |
| 22:24.46 | vasanthv161 | tzafrir: I saw some documentation. But all the terminology there didnt make much sense to me. Is a 'soft phone' the same as a SIP client? |
| 22:25.19 | tzafrir | yup. It is a soft[ware-implemented] [(normally SIP)] phone |
| 22:25.53 | vasanthv161 | So, in the link I provided above, the documentation connects two SIP phones. I can use two Google Talk sessions to accomplish the same? |
| 22:26.04 | vasanthv161 | (or maybe use the ones in Ubuntu that you mentioned) |
| 22:26.19 | tzafrir | I would suggest that you start with a local SIP phone. |
| 22:26.35 | tzafrir | It's easier to debug when you don't have to depend on a remote provider |
| 22:27.13 | vasanthv161 | What is a local SIP phone? :-s |
| 22:27.22 | vasanthv161 | You mean a hardware sip phone? |
| 22:28.09 | tzafrir | for gtalk you should probably use chan_gtalk (or is it chan_jingle?) |
| 22:28.34 | tzafrir | a local SIP phone: a software like Ekiga and Twinkle :-) |
| 22:29.03 | vasanthv161 | ah, ok. Let me look into that then. I will set up two softphones, and see if I can get them to work first. |
| 22:29.18 | vasanthv161 | Are you, by chance, familiar with tomato and running tomato on a router? |
| 22:30.38 | vasanthv161 | My eventual goal is to run asterisk on a router, so that i dont necessarily have to have my desktop running, for telephony. |
| 22:31.39 | tzafrir | How much memory and "disk" space do you have on your router? |
| 22:32.14 | tzafrir | IIRC openembedded have a package of Asterisk, but I'm not sure how up-to-date it is |
| 22:32.27 | vasanthv161 | I am not there yet. But I think I read somewhere that if I get a router that has a USB port, I can have a USB flash drive to run asterisk |
| 22:32.34 | tzafrir | If you run debian/Gentoo/whatever you'll probably be able to use the existing package |
| 22:33.20 | vasanthv161 | Anyway, that is a long way away from where I am now. Let me get some basic stuff working first. Thanks for the pointers. |
| 22:34.12 | tzafrir | For starters, stick with the default dialplan (maybe unload pbx_ael2.so and pbx_lua.so to get something cleaner) |
| 22:34.40 | tzafrir | See the sample sip.conf to see how to configure a SIP phone |
| 22:35.05 | tzafrir | also: copies of sample config files are under /usr/share/doc/asterisk-doc/examples/configs |
| 22:35.14 | vasanthv161 | Yeah. There are some seemingly good instructions in the asterisk docs. I was kinda stuck at using a softphone. Now I know where to look. |
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| 23:12.37 | tash | Is there a better sound file type than other? Wondering if I should use .wav, .gsm, or something else...? |
| 23:14.14 | tash | I'm currently creating files with the swift command, as .wav's...and converting them to gsm with "sox file.wav -r 8000 file.gsm" and putting the .gsm in my sound file directory. It doesn't sound like the best quality though. |
| 23:25.18 | WIMPy | Use the same format as you use for calls. |
| 23:26.02 | WIMPy | If you use multiple or are unsure, use wav with channel of 16 bit slin, 8k samples/s. |
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| 23:47.52 | tash | WIMPy, thanks. |
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