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20:15.52 | dan_j | Hi. Is anyone experiencing problems with app_voicemail in v13? I'm recording my own greetings using an agi script and it hangs Voicemail() when trying to play the greeting. So I wrote my own Voicemail() agi to handle people leaving messages. But now VoicemailMain() hangs when it tries to play back the messages. |
20:16.11 | dan_j | My only choice at the moment is to write another agi script to replicate VoicemailMain() functionality. |
20:17.06 | igcewieling | don't forget to mention you using database storage. |
20:17.55 | igcewieling | On non-realtime systems I'd make sure the msgxxxx.txt is correct if I saw that issue. |
20:18.31 | dan_j | I'm using realtime. It works totally fine in v11, but the exact same code in v13 causes a deadlock |
20:19.05 | igcewieling | dan_j: nothing useful in the UPGRADExx.txt files |
20:19.16 | dan_j | I'll double check but I dont think so. |
20:19.55 | igcewieling | "changes which break things" should be documented there. |
20:21.15 | igcewieling | also enable query logging on your database (mysql?), that helped me a lot when I was playing around with Realtime. |
20:22.28 | igcewieling | Eventually I realized it was just as easy to export the realtime sip configs into a config file, which gave me the functionality for Realtime caching without all the problems with realtime caching. |
20:23.07 | dan_j | Yes, I've also moved my sip peers out of realtime into static files which get automatically re-written if necessary and reloaded. |
20:23.39 | dan_j | No vm breaking changes listed there. I think it might be some sort of bug but since asterisk is just deadlocking, there is no obvious output to report |
20:24.27 | dan_j | Looks like I'm spending the evening writing an agi version of VoicemailMain :( |
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21:12.04 | mknooihuisen | Hi all. This may seem a very basic question, but I am a programmer and I've been doing a lot of research recently on using asterisk for my business. The one thing I cannot figure out, which seems crucial to getting started is: Where do I get a (preferably toll-free) phone number to use with asterisk? |
21:15.03 | WIMPy | ~itsp |
21:15.03 | 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. |
21:16.14 | mknooihuisen | Thanks WIMPy. :) |
21:16.42 | mknooihuisen | ~itsplist-us mknooihuisen |
21:17.10 | mknooihuisen | well, I tried to get it to send it just to me :P |
21:17.17 | mknooihuisen | ~itsplist-us |
21:17.17 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
21:17.34 | WIMPy | /msg infobot |
21:18.08 | mknooihuisen | Of course! I used to run an IRC server, but it's been years. |
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22:58.28 | mknooihuisen | Okay, quick follow-up to my original question about phone numbers: I'm looking at sipstation.com and thinking that I only need a DID or Toll-Free number (not a High Volume two-way trunk) since I'm planning on routing calls through Asterisk myself. Do I have this correct? |
23:01.10 | drmessano | mknooihuisen: Kinda |
23:01.21 | drmessano | mknooihuisen: But alos, kinda not |
23:01.34 | drmessano | The "Trunk" is a fixed $24.99 per one channel |
23:01.43 | drmessano | Without that, you're paying 2.4 cents per minute |
23:01.55 | drmessano | Has nothing to do with Asterisk.. it's 2 different payment options |
23:03.07 | drmessano | I'll go ahead and recommend a couple things |
23:03.13 | drmessano | and then others can argue it out |
23:03.16 | mknooihuisen | Yes, please do |
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23:04.22 | drmessano | Flat rate plans are never unlimited. $24.99 will buy you an unlimited* channel, but only up to a certain number of minutes. If you do the math, per minute, you're better paying per minute.. Having said that.. |
23:05.03 | drmessano | I would recommend someone completely different. Flowroute is inexpensive, high quality, and they don't try to force you into limited "unlimited" plans |
23:05.38 | mknooihuisen | Alright, I was just going down the list at itsplist-us |
23:05.43 | drmessano | Expect to pay about a penny per minute |
23:06.39 | mknooihuisen | I didn't mean to interrupt, by the way, in case you have more advice |
23:06.45 | mknooihuisen | :) |
23:11.26 | drmessano | Nope |
23:11.44 | drmessano | ITSP's are like religions.. or Linux distros. Highly contested |
23:11.47 | Samot | Most unlimited plans have a "Fair Use" policy. |
23:11.52 | drmessano | But Flowroute is solid |
23:11.52 | Samot | Which is different per provider. |
23:11.57 | drmessano | Right |
23:12.39 | Samot | If you abuse that policy it generally states they can push you to per minute plan. |
23:12.51 | Samot | Or just stop your service. |
23:13.27 | drmessano | $24.99 will usually mean like 3000 minutes.. which by the time you pay taxes and whatever else they tag on, is $30.00 .. which is $.01 per minute with a decent, non-shitty provider |
23:13.54 | Samot | Well I have my own opinions on SIPStation. |
23:14.02 | drmessano | Yes. Theres that too |
23:14.13 | drmessano | But I was trying to be objective |
23:14.25 | mknooihuisen | I'm looking through flowroute now and I'm curious if anyone knows anything about outbound calls, such as for a call center or customer service center situation |
23:14.27 | drmessano | The math doesn't work out |
23:14.37 | drmessano | What do you want to know |
23:14.46 | drmessano | Dont ask to ask, ask |
23:15.15 | mknooihuisen | drmessano, sorry, thought I was.... they don't seem to support them |
23:15.22 | drmessano | ROFL |
23:15.25 | mknooihuisen | just inbound calls as far as I can see :P |
23:15.41 | drmessano | Flowroute is a big provider to call centers |
23:15.59 | drmessano | Not sure what you're looking at.. they move lots of call center traffic |
23:16.23 | drmessano | https://www.flowroute.com/pricing/voice/ |
23:17.38 | drmessano | I've clicked 3 places where they show outbound calls |
23:17.46 | mknooihuisen | Sorry, it's probably just that I'm not understanding what I'm seeing. All the documentation and tutorials I've looked at have dealt with routing calls, dialplan and ARI, nothing has really covered actually getting the number. |
23:18.34 | drmessano | Sign up.. Buy a DID.. Setup your SIP credentials in Asterisk.. Start making calls |
23:21.27 | mknooihuisen | Sounds simple enough. It must be able to handle concurrent calls through 1 DID if call centers can use it. Correct? |
23:22.39 | mknooihuisen | drmessano, Oh wow, "Outbound SIP Trunkng" I'm not blind... at least not usually. I promise. :P |
23:24.05 | Samot | Flowroute doesnt limit channels. |
23:24.07 | drmessano | DIDs are just numbers. |
23:24.19 | Samot | You get 10 calls happening...you pay for usage. |
23:24.23 | drmessano | You Can make as many concurrent calls as you want |
23:24.28 | Samot | In or out. |
23:25.10 | drmessano | You can have 2 million concurrent calls to your DID |
23:25.16 | Samot | Lol |
23:25.36 | Samot | Ok Dr. Evil. |
23:26.06 | Samot | "How many calls do you do?" |
23:26.08 | drmessano | Only time a DID really comes into play with outbound calls is that you should set a CID on your calls. |
23:26.17 | Samot | ONE MILLLLLIIIOOONNN. |
23:26.24 | drmessano | And it be a number that exists |
23:26.41 | drmessano | But DIDs have nothing to do with outbound calls |
23:27.36 | drmessano | Forget the notion of analog lines and numbers. DIDs are just like IP addresses. You can have 1 or 1 million all pointing to the same place. Has no bearing on actual connections |
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23:30.27 | Samot | Even a DID is made up three parts. Kinda like a domain. |
23:30.47 | Samot | Well a NA based one. |
23:32.34 | mknooihuisen | Awesome. Makes me wonder why phone providers are so darn expensive. |
23:34.25 | Samot | Lots of wires. |
23:34.56 | mknooihuisen | Hah |
23:36.58 | Samot | Well look at it this way.. |
23:37.18 | Samot | In North America there are 9 major geographically regions. |
23:37.38 | Samot | In side those there are multiple LATAs to cover sections of those regions.. |
23:37.56 | Samot | Each LATA has multiple Central Offices... |
23:38.15 | drmessano | Youre paying for copper, not calls, basically |
23:38.19 | drmessano | and regulation fees |
23:38.26 | Samot | Each Central Office services 10,000+ subscribers. |
23:38.35 | Samot | And interconnections. |
23:39.18 | Samot | Oh yeah |
23:41.02 | Samot | Even when wireless becomes a standard for data delivery, the PSTN will still have copper. |
23:41.17 | Samot | There will always be some sort of physical wires. |
23:46.21 | drmessano | Not running to each house |
23:46.48 | drmessano | Those billions of miles of legacy copper are why AT&T and Verizon continue to charge so much for a land line |
23:48.07 | Samot | That's what I mean |
23:48.22 | Samot | Even if every house was wireless for data, the PSTN still will have copper. |
23:48.36 | Samot | If if all home/business phone service was VoIP. |
23:50.02 | mknooihuisen | I suppose the one advantage is that power outages do not necessarily impact phones |
23:50.53 | mknooihuisen | which, of course, they would with a VoIP system. |
23:51.01 | Samot | That's because POTS need voltage to handle calls. |
23:51.09 | drmessano | LOL |
23:51.20 | drmessano | That's no longer accurate |
23:51.28 | Samot | Well probably not anymore. |
23:51.46 | Samot | But they used to have a small charge. |
23:52.17 | drmessano | Years ago, you had battery voltage from CO to customer. Now you're relying on AC power at the cabinet, which may or may not have a working backup or one at all |
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23:52.33 | Samot | True. |
23:53.07 | Samot | But really.. |
23:53.17 | Samot | Even if your VoIP phone goes down these days.. |
23:53.22 | Samot | You pull out your cell phone. |
23:53.27 | drmessano | Battery from the CO is no longer a reality.. So VoIP relying on whatever your connectivity is, consider it no more fragile. |
23:53.38 | drmessano | and really |
23:54.01 | drmessano | If AT&T loses their fiber to the cabinet, your AT&T fiber for internet won't be up either. Same for the cable company, etc |
23:54.26 | Samot | Yup. |
23:54.31 | mknooihuisen | True enough |
23:54.33 | Samot | Like I said, cell phone. |
23:54.37 | drmessano | Basically there's very little chance AT&T or Verizon landlines will be up if you're not |
23:54.44 | Samot | Take VoIP out of the world. |
23:54.53 | Samot | Cell phone would still be the backup answer. |
23:54.56 | Samot | Because we all have them. |
23:55.41 | mknooihuisen | Yeah, although there are situations where cell coverage is weak or non-existent. |
23:56.01 | Samot | No situation is 100% |
23:56.14 | Samot | These would still be conversions if POTS was the only thing. |
23:56.27 | mknooihuisen | Sure |
23:56.40 | Samot | It's just easier to cling on to the 10 year old arguments about VoIP. |
23:57.16 | Samot | But like drmessano says... |
23:57.25 | Samot | People getting "POTS" these days are getting VoIP. |
23:57.42 | Samot | They just don't care/know because it's still their cordless phone at the house. |
23:57.55 | Samot | "I still plug it into my wall jack, so regular service" |
23:58.09 | drmessano | I tell people all the time |
23:58.19 | drmessano | If you want it to feel like POTS from the 70s |
23:58.21 | drmessano | Buy a UPS |
23:58.25 | drmessano | Bam, Battery |
23:58.29 | drmessano | But |
23:58.31 | Samot | ^ Bingo |
23:58.43 | drmessano | Thats only as good as your house |
23:58.50 | drmessano | Hope AT&T has been maintaining generators |
23:59.08 | Samot | oh they are. |
23:59.19 | drmessano | Yeah |
23:59.24 | Samot | The CO across from our data center at the CLEC.. |
23:59.26 | drmessano | Everytime the power goes out |
23:59.28 | Samot | Had 4 generators. |
23:59.32 | drmessano | They send out a truck |
23:59.45 | Samot | Well even we had two generators at the DC. |
23:59.50 | drmessano | I'm not talking about at the CO |