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| 00:39.58 | mojtaba | Hi, I am new to Asterisk. Do you know any resource to configure asterisk? |
| 00:40.11 | mojtaba | I just want to connect it to a PSTN. |
| 00:40.23 | mojtaba | I have installed it on a raspberry pi. |
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| 02:52.42 | tonsofpcs | lvlinux: I got it working finally with ffmpeg ... now to make it actually be the on-hold destination for this phone |
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| 05:37.56 | Lope | what are your thoughts on running a self-compiled, latest asterisk vs the latest release distributed by a distro? |
| 05:42.48 | ChannelZ | I've never run a distro version |
| 05:43.18 | ChannelZ | I've compiled myself since I started using asterisk on 1.4 |
| 05:44.04 | Rasputin3711 | ChannelZ: What version are you using now for production? |
| 05:44.39 | ChannelZ | I'm on 13.7 |
| 05:45.00 | Rasputin3711 | How many users per server? |
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| 05:46.12 | ChannelZ | It's my office system with a dozen phones.. plus another system at home |
| 05:46.53 | ChannelZ | I'm not a call center or anything. But compiled or distro shouldn't make a damn bit of difference |
| 05:47.22 | Lope | ChannelZ: how often do you recompile? |
| 05:48.09 | Lope | ChannelZ: don't you have concerns that you might be sitting on 13.7 and then an exploit comes out for it that you're unaware of, 13.8 gets released but you're busy etc and your stuff gets hacked? |
| 05:48.20 | ChannelZ | eh, maybe every few months.. just depends on what's going on, both with me and asterisk. I'm obviously a few versions behind on my work system. |
| 05:48.47 | ChannelZ | I've generally not personally hit any bugs I've HAD to upgrade to fix |
| 05:49.04 | Lope | ok |
| 05:49.33 | ChannelZ | And I keep up on security stuff so I'd generally be aware if something was discovered and I needed to do something about it |
| 05:49.43 | Lope | 2. Do you guys have any thoughts of a Realtime linux kernel for asterisk, like if there's any need/benefit? |
| 05:49.58 | Lope | 3. Do you have any preferences for using a non-standard IO scheduler? |
| 05:51.02 | ChannelZ | Can't help you there, my systems are nowhere near the size for it to matter. |
| 05:51.49 | Lope | oh, okay. I'm experiencing frequent jitter on my digitalocean VPS. So I'm going to try run it on one of my servers and see if there's a diff. |
| 05:52.23 | Lope | The only thing I could really do on my VPS was nice asterisk ro -21 but it didn't make any difference. |
| 05:52.33 | Lope | s/ro/to |
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| 05:54.10 | ChannelZ | Is it a private virtual server or there's any number of other users/VMs on the same hardware? |
| 05:55.23 | Lope | it's a VPS, they likely have many other users sharing the same hardware. |
| 05:55.37 | Lope | They don't advertise their contention ratio. |
| 05:56.04 | Lope | So if it works on my own server, I'll try another host. |
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| 05:56.20 | ChannelZ | Well then unless you're running tons of other stuff within your own VM I'm not sure how nice or anything else would alleviate any load problems if there are other VMs outside yours which you have no control over |
| 05:57.34 | Lope | Yes exactly. So it's pretty definite that alternative hosting is the only solution. If infact it's the cause of the jitter I experience regularly. |
| 05:58.13 | ChannelZ | And an RT kernel is going to just increase the frequency of interrupts which actually increases load somewhat so within a VM I'm not sure that will be of help either. But I'm not a kernel nerd, just thinking about it logically. Someone else might have more insight |
| 05:59.03 | ChannelZ | Also do you know the jitter is CPU related and not network? |
| 05:59.22 | Lope | yeah since my VM is idling aside from asterisk I don't think there's anything I can do to alleviate it. The problem is at the level above. |
| 05:59.49 | Lope | ChannelZ: it could be network as well. But in that case I also have no control over it. |
| 06:00.23 | Lope | They've obviously got a lot of servers in a small space, and they have to schedule the network packets somehow. |
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| 06:01.02 | ChannelZ | Yeah. There's several variables in a hosted situation that could be difficult if not impossible to narrow down for sure. Whether it's the host hardware being overloaded and your VM being starved, or the network bandwidth to the host hardware at capacity, or a network connection from the hosting facility to the rest of the world issue.. |
| 06:01.17 | Lope | Does asterisk have any related benchmarking software that can test a host's eligibility? |
| 06:02.04 | Lope | I've tried setting the jitter buffer at a max of 2000 but the audio I hear from my server when I call it, is all jittered up. |
| 06:02.18 | Lope | sorry I meant to say a min of 2000ms and a max of 5000ms |
| 06:02.38 | Lope | Also tried adaptive with a default of 2000ms. No diff there either. |
| 06:02.53 | ChannelZ | well there's 3rd party SIP load testers that can barrage your system with calls and try to figure out capacity but in your case you really need to know what the host hardware is doing, you already know the VM is limping |
| 06:02.55 | drmessano | What size VPS? |
| 06:03.00 | Lope | Since I've tried the jitter buffer, it's probably more likely to be a network problem. |
| 06:03.05 | Lope | 512mb RAM 15GB |
| 06:03.09 | drmessano | ..... |
| 06:03.12 | ChannelZ | Jitter buffers only affect the receiving end |
| 06:03.12 | drmessano | Yeah no |
| 06:03.20 | drmessano | That VPS will never work |
| 06:03.22 | ChannelZ | IE you |
| 06:03.47 | Lope | It's only a tiny extensions.conf I'm running, with like 3 little extensions, and I'm only handling 1 call at a time in testing, and getting jitter? |
| 06:03.58 | drmessano | I never recommend the $5 DO VPS for Asterisk, ever |
| 06:04.02 | ChannelZ | You buffer a certain amount of packets you receive to smooth out packet latency.. but what you send out is what you send out. |
| 06:04.05 | drmessano | Get the next one up |
| 06:04.25 | Rasputin3711 | drmessano: What version are you using now for production? |
| 06:04.25 | Lope | drmessano: have you tested? |
| 06:04.35 | drmessano | Yes I have |
| 06:04.46 | drmessano | Why do you think I know so much about it? |
| 06:05.02 | drmessano | The $5 VPS is good for like a web host |
| 06:05.13 | drmessano | Not for a realtime system |
| 06:05.24 | drmessano | You need to move up a tier |
| 06:05.35 | Lope | $5: 0.5GB 1core 20GB SSD. $10 1GB 1core 20GB SSD. Both have 1core? |
| 06:05.54 | drmessano | The $10 one with a GB of RAM is fine for a small system |
| 06:08.11 | Lope | `free -h` Tot:490M. used:467M, Free:22M, buffers:68M, cached:260M |
| 06:08.18 | drmessano | .... |
| 06:08.26 | drmessano | Ok, this isnt about Asterisk dimensioning |
| 06:08.33 | drmessano | or memory usage |
| 06:08.43 | Lope | okay? |
| 06:08.57 | drmessano | A DIGITAL OCEAN $5 VPS PERFORMS POORLY |
| 06:09.09 | Lope | So you're saying the bigger VM's run with less CPU contention? |
| 06:09.11 | drmessano | It's not about advertised specs |
| 06:09.18 | drmessano | Get the $10 one |
| 06:09.38 | Lope | okay, will try diff sizes and at diff hosts |
| 06:09.56 | Lope | first, I've got my own server, so I know it's not too busy, will try there. |
| 06:10.08 | Lope | but it's far away from where I need the server to be situated. |
| 06:10.24 | Lope | latency wise. |
| 06:10.25 | drmessano | So get the $10 DO option |
| 06:10.29 | drmessano | As I keep suggesting |
| 06:10.35 | drmessano | or a $10 Linode |
| 06:10.45 | drmessano | Either of those is fine for a small system |
| 06:11.06 | Lope | ok, thanks for the recommendation, I've noted :) |
| 06:12.44 | drmessano | Generally the cheap ass VPS'es are all shoved on the same overcrowded boxes with users trying to squeeze everythng they can out of $5 or $7 |
| 06:13.19 | drmessano | So I find that while the specs look good, and Asterisk should be fine, the VMs are highly latent |
| 06:14.17 | Lope | yeah |
| 06:14.30 | Lope | Makes sense. |
| 06:14.36 | drmessano | Probably every VM on that host is being pushed hard by cheap asses |
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| 06:29.25 | Samot | Word. |
| 06:34.27 | ikkuranus | could someone help me diagnose why asterisk wont autostart on my pogoplug? |
| 06:36.34 | ikkuranus | manual start does work |
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| 06:59.20 | ikkuranus | the following exists: /etc/init.d/asterisk and has the right permissions |
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| 07:16.08 | ikkuranus | eh nevermind seems to be starting just takes forever |
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| 09:50.46 | dokma | Bloody hell, the amount of acronyms in the PBX area is killng me! |
| 09:51.13 | dokma | All I want is to plug my PSTN chord into my PC and be able to talk using my speakers and mic. What do I need for that? |
| 09:51.38 | dokma | I'm using Debian GNU/Linux. |
| 09:53.05 | Samot | Why wouldn't you use a softphone client to connect to your PBX? |
| 09:54.12 | dokma | I don't have a PBX. All I have is a PC with Debian on it and a PSTN plug in the wall. |
| 09:54.44 | dokma | Obviously I need some peace of hardware in my PC to make it happen and probably some software on my Linux machine. |
| 09:54.52 | Samot | Yes. |
| 09:55.02 | dokma | So I'm trying to navigate the confusing land of this whole thing. |
| 09:55.03 | Samot | You need an FXO card and Asterisk. |
| 09:55.09 | dokma | What do I need? IN SIMPLE TERMS! |
| 09:55.16 | Samot | You need a server. |
| 09:55.21 | dokma | Ah, that's a good answer. |
| 09:55.24 | Samot | That has an FXO PCI card. |
| 09:55.34 | Samot | Linux OS |
| 09:55.42 | Samot | And Asterisk. |
| 09:55.42 | dokma | Awesome, that IS a helpful answer. |
| 09:56.05 | dokma | So now I have to find a list of Linux compatible FXO cards. |
| 09:56.13 | Samot | Or you can get an external FXO device that connects to Asterisk via SIP. |
| 09:56.15 | Rasputin3711 | simple voip gateway |
| 09:56.18 | Samot | No. |
| 09:56.33 | Samot | You would need to find a list of FXO cards that Asterisk can use. |
| 09:56.49 | dokma | Asterisk and Linux I guess. |
| 09:56.51 | Samot | Like the Sangoma. |
| 09:57.06 | Samot | But there are external options as well. |
| 09:57.12 | Samot | That require less configuration. |
| 09:57.30 | Samot | And pulling your hair out getting the drivers to work. |
| 09:57.35 | dokma | Can't find a used or new Sangoma in my country. |
| 09:57.55 | Samot | Well they are one brand. |
| 09:58.10 | Samot | dokma: How much experience do you have with Linux? |
| 09:58.15 | dokma | Let me search for used FXO cards on the leading small ads portal in my country. |
| 09:58.29 | dokma | 15 years of Linux only administration. |
| 09:58.32 | Samot | OK. |
| 09:58.54 | Samot | This just for home use? |
| 09:59.13 | dokma | For a business that is in a home ATM. |
| 09:59.32 | dokma | I find lot's of external devices when I search FXO and no internal PCI cards. |
| 09:59.57 | Samot | I'd go with an external one if it's easier for you to get. |
| 10:00.13 | Samot | Because it will use SIP to connect back to the PBX. |
| 10:00.50 | dokma | Patton SN4114 4-FXS VoIP Gateway SN4114/JS/EUI |
| 10:00.57 | Samot | No. |
| 10:01.06 | Samot | FXS is not the same. |
| 10:01.18 | Samot | Your walljack is FXS. |
| 10:01.37 | Samot | The jack on the back of your analog phone/fax machine, that's FXO. |
| 10:01.37 | dokma | It does have FXO interface |
| 10:01.42 | Samot | K. |
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| 10:02.00 | dokma | So I have an FXS on the wall right? |
| 10:02.05 | Samot | Yes. |
| 10:02.11 | dokma | And I need to plug the device I'm about to buy into FXS. |
| 10:02.11 | Samot | Most simple way to think of it: |
| 10:02.15 | Samot | FXS = Out |
| 10:02.20 | Samot | FXO = In. |
| 10:02.41 | dokma | So the wall has FXS and the device will have FXO and those two will be connected? |
| 10:02.47 | Samot | Right. |
| 10:02.50 | Lope | I want to run asterisk behind nat. this should be pretty easy, just forwarding 5060 to my asterisk LAN IP and a range of RTP ports for SIP. However then do I need to tell asterisk what it's internet IP is? |
| 10:02.57 | Samot | Just think of the device as a regular phone. |
| 10:03.09 | Samot | You take the phone cord and plug it into the phone and into the wall. |
| 10:03.13 | dokma | That's awesome analogy. Helps A LOT! |
| 10:03.28 | dokma | So my ordinary phone has an FXS plug in it as well... |
| 10:03.38 | dokma | I'm learning finally... |
| 10:03.47 | dokma | MEDIATRIX C710 |
| 10:04.09 | dokma | That is my phone has an FXO in it... |
| 10:04.21 | Samot | All analog phones are FXO. |
| 10:04.25 | Lope | ah externip=1.1.1.1 and localnet=192.168.1.0/24/255.255.255.0 |
| 10:04.32 | Samot | All analog fax machines are FXO. |
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| 10:05.55 | Samot | Lope: Those would both be wrong. |
| 10:06.08 | Samot | localnet=192.168.1.0/24 |
| 10:06.10 | Samot | That's it. |
| 10:06.37 | Samot | externip=<WAN IP> |
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| 10:17.45 | dokma | Seems I lost my conn... |
| 10:18.32 | dokma | So one more question from me. When I connect my PSTN to my PBX device will I be able to have an incoming call received on my Laptop via a VPN? |
| 10:18.38 | *** join/#asterisk areski (~areski@120.red-88-27-170.staticip.rima-tde.net) |
| 10:29.01 | Samot | dokma: You would use a softphone client like X-Lite or Zopier. |
| 10:29.15 | Samot | If you want to reach it remotely when you're not home you can setup NAT rules for that. |
| 10:29.20 | dokma | Samot, so it is possible. |
| 10:29.31 | dokma | Samot for remote I use VPN |
| 10:29.58 | dokma | What is the correct name for these devices that bridge PSTN to my PC? |
| 10:30.12 | Samot | softphone client. |
| 10:30.48 | Samot | There is nothing on your PC that lets you handle a phone jack connection. |
| 10:31.10 | Samot | For a pc you would use a softphone. |
| 10:31.13 | Rasputin3711 | Samot: What version are you using in production? |
| 10:31.14 | dokma | So a softphone client is the thing that I plug the cable from my wall FXS jack into? |
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| 10:32.48 | Samot | dokma: No. |
| 10:33.20 | Samot | Rasputin3711: I try to keep all my systems on the latest version. |
| 10:33.26 | Samot | So 13.9.1 |
| 10:33.27 | dokma | Samot: what is the correct name for the thing I plug the cable from wall FXS into that converts analog into digital signal? |
| 10:34.06 | Samot | FXO gateway |
| 10:34.17 | Samot | I thought we covered this. |
| 10:34.37 | dokma | We did but the haze in my head from all these acronyms hasn't settled yet. |
| 10:34.55 | dokma | I'm building the grand picture so stick with me... |
| 10:35.52 | dokma | So once an FXO gateway is all set and working I can plug a PC into it with a network card or a WIFI conn and use a softphone client on the PC? |
| 10:36.07 | Rasputin3711 | Samot: How maximum users on one server have you had? |
| 10:36.09 | dokma | Or I can plug a Skype phone directly into the FXO gateway? |
| 10:36.57 | Samot | dokma: What is the absolute end result you are looking for? |
| 10:37.06 | Rasputin3711 | http://www.3cx.com/pbx/fxs-fxo/ |
| 10:37.11 | Samot | Rasputin3711: That goes for you too. |
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| 10:37.24 | dokma | Samot: basically to be able to use all my PCs and laptops as phones. |
| 10:37.39 | Samot | Rasputin3711: You've been asking people a lot of questions about their deployments. What are YOU looking to do? |
| 10:37.57 | Rasputin3711 | Samot: Best practices |
| 10:38.13 | Samot | dokma: Then you need to either get a VoIP service that you can connect softphone clients to. |
| 10:38.42 | Samot | dokma: Or you need to get a PBX like Asterisk to connect to the PSTN and then connect your softphone clients too. |
| 10:38.52 | dokma | I'm opting for #2 |
| 10:38.59 | Samot | dokma: Your PCs and laptops will run softphone clients. |
| 10:39.19 | Samot | dokma: #2 is a lot more work. |
| 10:39.37 | dokma | How much more? |
| 10:39.42 | Samot | Rasputin3711: Best practices for what? |
| 10:40.04 | Rasputin3711 | Samot: users,hardware,software version |
| 10:40.21 | Samot | dokma: Well you need to build the server, get the FXO device, install Asterisk, configure both Asterisk and the FXO device, then configure the softphones. |
| 10:40.38 | Samot | Rasputin3711: For what?! Best practices for what? |
| 10:41.02 | Samot | Best practices for how I use Asterisk as gateways for my voice network? |
| 10:41.13 | Samot | How I use it for individual PBX deployments? |
| 10:41.20 | Rasputin3711 | For many cases |
| 10:41.30 | Samot | google.com |
| 10:41.34 | Rasputin3711 | For small and medium |
| 10:42.08 | dokma | Samot, I think I can handle that no problem once I build the whole general landscape of this whole nitche. |
| 10:42.27 | Samot | Users, software and hardware is just a part of it all, Rasputin3711. |
| 10:42.29 | dokma | Samot: btw, what is the name of this whole general nitche? Is it VOIP, PBX? |
| 10:42.41 | Samot | Telephony. |
| 10:43.01 | dokma | Just Telephony? I would confuse that with PSTN... |
| 10:43.14 | Samot | Public Switched Telephone Network. |
| 10:43.25 | Samot | IE. how all the carriers in the world talk to each other. |
| 10:43.48 | dokma | That's the plain old PSTN we all know of. But when you throw PCs and networks into the mix doesn't that have some distinguishing name? |
| 10:43.56 | dokma | Like IP Telephony or something |
| 10:44.00 | Samot | Sure. |
| 10:44.04 | Samot | But the PSTN is the PSTN |
| 10:44.41 | dokma | I get that. But what is all this we were talking about... gateways, VOIP providers, PBXes etc... |
| 10:44.52 | dokma | Still just telephony? |
| 10:44.56 | Samot | PBX hasn't changed. |
| 10:45.03 | Samot | A PBX is a PBX. |
| 10:45.23 | Samot | You're either using TDM aka copper or you're using IP. |
| 10:45.31 | dokma | No I get that. I was looking for a covering term that encompasses al that... |
| 10:45.43 | Samot | Telecommunications |
| 10:45.43 | dokma | So it is IP Telephony right? |
| 10:45.49 | TandyUK | dokma: its to do with making telephone calls, that makes it al ltelephony, regardless of which specific technology (PSTN, ISDN, VOIP, Mobile) youre using |
| 10:46.05 | TandyUK | ^^^ |
| 10:46.15 | Samot | PSTN is involved in ALL of it. |
| 10:46.23 | dokma | Hmmm... ok. That's why it's all hazy to me. I don't have a basic coverage of the terms... |
| 10:46.27 | Samot | Public Switched Telephone Network. |
| 10:47.06 | dokma | Yes, I get it now. I thought that PSTN is like "old stuff" and this new stuff with networks etc was "IP Telephony" |
| 10:47.13 | Samot | Conceptually it's no different than how Internet backbones work. |
| 10:47.19 | TandyUK | dont forget PATS too (Publicly Available Telephone Service), thats what emcompasses the requirement for you to be able to phone '999' for example |
| 10:47.22 | Samot | That's POTS |
| 10:47.32 | Samot | Plain Old Telephone Service. |
| 10:47.34 | Samot | Copper. |
| 10:47.45 | Samot | IP and Copper are just delivery methods. |
| 10:47.53 | Samot | That all connect to the PSTN. |
| 10:48.12 | dokma | Kewl. The picture is getting a bit clearer now... |
| 10:49.01 | dokma | One thing that is still rather hazy to me is when I look at used gateways I have no idea if a particular one is an FXO gateway. |
| 10:49.06 | Samot | You know when you do a network diagram of how all your office networks will interconnect and you put a big cloud in the center and call it "PUBLIC INTERNET" |
| 10:49.23 | Samot | You would do the same thing with your phone network and change "PUBLIC INTERNET" to "PSTN" |
| 10:49.24 | dokma | What should a gateway have listed to be certain it can do the PBX thing? |
| 10:49.48 | Samot | If it's an FXO gateway it will say it's an FXO gateway. |
| 10:49.54 | dokma | Samot: yes, that is an AWESOME analogy. A workable one that helps. |
| 10:50.25 | dokma | Can you take a peak at this one: http://www.njuskalo.hr/wireless-wlan/bezicni-zicni-modem-voip-router-smc-barricade-7908-vowbrb-eu-v-2-oglas-10443814 |
| 10:50.33 | TandyUK | FXS: Connects to the phone line going out of the building, FXO: Connects to a device, such as phone or fax machine |
| 10:50.42 | dokma | It does not specifically say it's an FXO gateway... |
| 10:51.11 | TandyUK | Telefonija suÄelja: 1. telefon (FXS), 1 red ( FXO) |
| 10:51.22 | TandyUK | not sure what that means, but it has 1 fxs port and 1 fxo |
| 10:51.28 | dokma | Yes it has an FXS and FXO, I've seen that. |
| 10:51.39 | Samot | It means... |
| 10:51.41 | dokma | The "Telefonija suÄelja" means "The telephony of the interface" |
| 10:51.50 | Samot | That if it has one FXS and one FXO... |
| 10:52.09 | Samot | You can use it to connect to the POTS and the PBX as a failover. |
| 10:52.21 | dokma | So it does have those interfaces. BUT can it convert analog telephone signal to digital and enable me to use Asterisk with it? |
| 10:52.40 | Samot | That's the whole POINT. |
| 10:52.49 | Samot | To convert TDM/Copper to IP. |
| 10:53.24 | dokma | Hmmm.... if that is so then I potentially already have PBX ability with my current ISP providers router/gateway... |
| 10:54.06 | Samot | You only need FXS/FXO if you are converting Analog to Digital. |
| 10:54.10 | dokma | Well ain't that funny. My router's web interface has a VOIP tab... |
| 10:54.19 | Samot | That doesn't mean anything. |
| 10:54.32 | Samot | That's for networking settings related to VoIP usage. |
| 10:55.05 | dokma | My phone line comes via that same router/gateway. |
| 10:55.08 | Samot | dokma: Let's break it down like this.. |
| 10:55.12 | Lope | Samot: okay, thanks, sorry I wrote that in a hurry, had food on the stove. |
| 10:55.17 | dokma | My plain old phone is plugged into that router. |
| 10:55.28 | Samot | Right because the modem is doing the conversion. |
| 10:55.51 | dokma | The modem is converting what to what? |
| 10:55.59 | Samot | SIP to Analog |
| 10:56.10 | Samot | So it's an FXS device. |
| 10:56.24 | dokma | Technicolor TC7200 |
| 10:56.29 | dokma | That's the modem. |
| 10:56.35 | Samot | No bearing. |
| 10:56.40 | Samot | Look it's very simple. |
| 10:56.49 | Samot | You have a PBX. |
| 10:57.03 | dokma | That's a useful answer. |
| 10:57.07 | Samot | Now, how do you want to connect that PBX so it can make and receive calls to the world... |
| 10:57.18 | Samot | You have two options: Copper or IP. |
| 10:58.10 | Samot | Which one are you going to use? |
| 10:58.15 | dokma | When you say just connect it's a bit confusing. It's already connected to my ISP where it has an Internet and Phone connections. So you must mean connect it to something on my side? |
| 10:58.41 | Samot | Leave your ISP out of it right now. |
| 10:58.43 | dokma | On my side I've already plugged a plain old phone into it. |
| 10:58.54 | Samot | How do you want your PBX to be able to make and receive calls? |
| 10:58.57 | dokma | Now I'd like to make call from my PC via that same modem. |
| 10:59.07 | dokma | From ALL of my machines. |
| 10:59.11 | Samot | FFS. |
| 10:59.12 | Samot | I know |
| 10:59.25 | Samot | I gave you the two options in order to do that. |
| 10:59.37 | Samot | You choose the "My own PBX" option. |
| 11:00.05 | dokma | Yes. What I had in my head when I said that is: |
| 11:00.07 | Samot | We already know the computers will use a softphone client. |
| 11:00.12 | Samot | So that's done. |
| 11:00.23 | Samot | Now, how does the PBX send those calls to the rest of the world? |
| 11:00.31 | Samot | Or receive calls from the rest of the world? |
| 11:00.36 | dokma | Install Asterisk on my Debian box and use a softphone client on it to make calls via that modem I already use. Is that correct? |
| 11:00.38 | Samot | Will it be over a copper line or IP? |
| 11:01.12 | Samot | You don't install Asterisk on your PC. |
| 11:01.19 | Samot | You need to have it on it's own server. |
| 11:01.30 | Samot | Your PBX should be just that, a PBX. |
| 11:01.36 | Samot | Not your workstation. |
| 11:01.47 | dokma | Asterisk will probably be on my workstation. |
| 11:01.51 | Samot | No. |
| 11:01.53 | dokma | If that's possible. |
| 11:02.09 | Samot | OK, I'm done. |
| 11:02.12 | Samot | You are not listening. |
| 11:02.19 | dokma | Pardon me. |
| 11:02.29 | dokma | Where do I need to put Asterisk? |
| 11:02.45 | Samot | On it's own server. |
| 11:02.53 | Samot | As I said like an hour ago. |
| 11:02.55 | dokma | What is the point of that? |
| 11:02.57 | Rasputin3711 | Ask your isp about voip services. |
| 11:03.08 | Samot | Because it's a PBX |
| 11:03.31 | Samot | It's going to want to do things on the server... |
| 11:03.48 | dokma | That's not very understandable. How do the fact that Asterisk is a PBX necessitates an exclusive machine for it? |
| 11:04.15 | Samot | Because Asterisk is going to write things all over the server. |
| 11:04.34 | dokma | Really? That sounds strange. |
| 11:04.35 | Samot | It's going to want to do things on the server and have the server configured a specific way. |
| 11:04.52 | Samot | Really? |
| 11:05.07 | Samot | Would you by a 3CX PBX and then want to run your web server on it? |
| 11:05.08 | Samot | No. |
| 11:05.19 | Samot | You wouldn't buy a PBX system and then install other stuff on it. |
| 11:05.22 | Samot | Because it's the PBX system. |
| 11:05.45 | dokma | Hang on. When you say a PBX system does that mean a PC with a net connection and Asterisk on it? |
| 11:05.46 | Samot | You're telling me in all the years of Linux administration you've need had a server be just one thing?! |
| 11:06.02 | dokma | Naturally I did. |
| 11:06.06 | Samot | OK |
| 11:06.18 | Samot | So why is this such a hard concept to grasp in regards to this? |
| 11:06.39 | dokma | It's not that hard to grasp. It's just strange that it's a must. |
| 11:06.53 | dokma | In all my years of Linux admin I used exclusive servers for all sorts of things. |
| 11:07.04 | dokma | BUT I also used all those services on one machine without problems. |
| 11:07.09 | Samot | Sure. |
| 11:07.17 | Samot | But did you use them all on your workstation? |
| 11:07.19 | dokma | So to say that Asterisk is somehow special is strange. |
| 11:07.30 | dokma | I use the all on my workstation now for web dev. |
| 11:07.30 | Samot | Run it on your workstation. |
| 11:07.51 | Samot | Have your phone service depend on your workstation always being online.. |
| 11:07.57 | Samot | Never down.. |
| 11:07.57 | dokma | I'm running Apache, MySQL, Postfix, Dovecot, PowerDNS and who knows what else on this machine. |
| 11:08.08 | Samot | And it's always running the needed services. |
| 11:08.19 | dokma | My Workstation is never off. |
| 11:08.22 | Samot | OK |
| 11:08.25 | Samot | then run it there. |
| 11:08.39 | Samot | When you see how your service is impacted.. |
| 11:08.47 | Samot | And it doesn't work right... |
| 11:08.54 | Samot | You'll be back at putting it on it's own system. |
| 11:09.03 | Samot | While interrupting your phone services. |
| 11:09.09 | dokma | No, I haven't decided to run it from my workstation. |
| 11:09.26 | dokma | It was just strange how you adamantly demanded that it MUST be ran exclusively. |
| 11:09.40 | Samot | It's called experience. |
| 11:09.46 | dokma | So it CAN be ran on a normal workstation but you RECOMMEND to run it on an exclusive machine. |
| 11:09.54 | dokma | You presented it a bit off. |
| 11:10.23 | dokma | You have to keep in mind that this is very hazy to me and I'm learning everything from scratch... |
| 11:11.17 | dokma | So now I have this: a PC with Asterisk on it connected to this Technicolor modem which apparently has the needed functionality to make calls from my workstation. Is that correct? |
| 11:11.35 | Rasputin3711 | yeastart soho + O2 module ) |
| 11:12.32 | Samot | dokma: No. |
| 11:14.38 | Samot | Well sure, if you're going to use an FXO card/device and just use your current phone line. |
| 11:15.55 | Samot | But you really, really should spend time on wiki.asterisk.org to read up and learn. |
| 11:16.19 | dokma | Yeah, I know. I exhausted you real well. You're tremendeously helpful. |
| 11:16.52 | dokma | So this modem that I have the Technicolor does not have the FXO gateway functionality so I need one extra piece of hardware... |
| 11:18.30 | dokma | SO the picture is something like: PC with an FXO gateway and Asterisk <---> Technicolor modem with an FXS jack |
| 11:19.12 | dokma | Or if the FXO gateway is external: PC with Asterisk <---> FXO gateway <---> Technicolor modem with FXS jack. |
| 11:24.44 | Samot | Correct. |
| 11:30.38 | dokma | Samot: OK. What confused me was seeing that VOIP tab in my modems web interface. I thought that perhaps the modem already had an FXO gateway functionality builtin. |
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| 13:05.25 | Eyel | Hi guys!! |
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| 13:51.15 | iBog | hello. can anyone recommend an audio conferencing service? |
| 13:51.54 | [TK]D-Fender | We would.... but we're all using our own PBX's to do this for free.... |
| 13:52.23 | lvlinux | :hifive [TK]D-Fender |
| 13:52.30 | lvlinux | lol |
| 13:52.31 | iBog | :) just thought you may now of some other services |
| 13:52.56 | [TK]D-Fender | https://www.google.ca/#q=phone+conferencing+service |
| 13:53.01 | [TK]D-Fender | Google knows TONS |
| 13:53.11 | lvlinux | iBog: ZipDX is good and highly regarded. |
| 13:54.02 | iBog | lvlinux thank you, I really appreciate it |
| 13:54.14 | iBog | [TK]D-Fender yes, I've been googling too |
| 13:54.43 | lvlinux | iBog: do you need video as well or just audio? WebRTC? |
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| 13:55.10 | iBog | lvlinux just audio. let people dial into a number from their phone |
| 13:55.16 | lvlinux | ah |
| 13:57.33 | lvlinux | iBog: just out of curiosity, why would you be looking at another service vs doing it yourself with Asterisk? |
| 14:00.20 | iBog | lvlinux I want something setup rther quickly. I dont want to get equipment to setup asterisk. |
| 14:01.51 | lvlinux | ah, I see. Well that's reasonable. I'm sure you can find a service that will meet your needs, or for that matter, probably a number of people here would be interested in setting up a server for you, or hosting the service for you. (myself included) |
| 14:02.50 | lvlinux | Otherwise you can't go wrong w ZipDX IMHO. |
| 14:02.59 | iBog | very kind of you. am interested in asterisk though |
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| 14:04.06 | [TK]D-Fender | ZipDZX: Basic Service Package: $0.08 (USD) per minute per participant inclusive of Federal taxes and fees (even the Federal Universal Service Fund) |
| 14:04.08 | iBog | if I used asterisk, do I need to get phone lines and modems? |
| 14:04.18 | lvlinux | :-D |
| 14:04.20 | [TK]D-Fender | I guess if you don't hav you own and maybe have a very temporary need.... |
| 14:04.29 | lvlinux | iBog: all you need is a computer and internet connection |
| 14:04.39 | lvlinux | [stable] internet connection |
| 14:04.44 | iBog | lvlinux good to know. then a plan with a SIP provider? |
| 14:04.48 | lvlinux | yes |
| 14:04.50 | [TK]D-Fender | iBog, No, * does not support regular "modems". there are cards for analog lines if tthat's what you actuall want to use, but there are many more choices |
| 14:05.17 | [TK]D-Fender | Analog lines, Digital (CAS/BRI/PRI), VoIP, GSM, etc |
| 14:05.32 | lvlinux | all those are options |
| 14:05.34 | [TK]D-Fender | that's the reason for the name.... |
| 14:05.35 | iBog | are there any flat rate SIP plans? |
| 14:05.40 | lvlinux | yes |
| 14:05.50 | lvlinux | well, yes and no |
| 14:05.57 | TandyUK | not from us, but some do plans |
| 14:05.59 | [TK]D-Fender | Many, but most have ToS including soft/hard caps, etc |
| 14:06.10 | TandyUK | I tend to find plans just work out more expensive than just paying for calls |
| 14:06.20 | lvlinux | for incoming there are "nominal" flat rate programs. Most have a minute/month limite though. |
| 14:06.30 | TandyUK | unless youre doing like 4000 minutes/mo, in whcih case TOS usually applies |
| 14:06.30 | lvlinux | TandyUK: I agree in many circumstances |
| 14:06.41 | TandyUK | so its still cheaper to just buy what you need |
| 14:06.49 | lvlinux | AnveoDirect has a 30000 minute cap |
| 14:06.59 | lvlinux | most others are 4000 or so |
| 14:06.59 | iBog | ok. will expand to researching this option too |
| 14:07.07 | TandyUK | a lot have caps on simultaneous calls too |
| 14:07.20 | lvlinux | varies, some do some don't |
| 14:07.21 | TandyUK | eg, you can have 30k minutes, but only have 2 calls in progress at a time |
| 14:07.23 | iBog | what are a few SIP providers to consider? |
| 14:07.38 | TandyUK | iBog theres hundreds |
| 14:07.40 | iBog | fyi, I'm in canada |
| 14:07.42 | TandyUK | if not thousands |
| 14:07.50 | iBog | TandyUK I get that |
| 14:07.51 | lvlinux | iBog: AnveoDirect, Flowroute, Vitelity, Callcentric, DIDLogic are my picks |
| 14:08.00 | iBog | lvlinux much appreciated |
| 14:08.01 | TandyUK | you want a provider with servers in canada |
| 14:08.08 | iBog | TandyUK yes |
| 14:08.09 | lvlinux | and voip.ms |
| 14:08.11 | [TK]D-Fender | ~itsplist-ca |
| 14:08.17 | infobot | hmm... itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 14:08.21 | [TK]D-Fender | ~itsplist-us |
| 14:08.21 | 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 |
| 14:08.41 | [TK]D-Fender | Hrm... some of those should be removed... |
| 14:08.45 | TandyUK | lol |
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| 14:11.10 | _AxS_ | Hey all -- is there bychance a good diy-pbx-for-dummies guide you all recommend? not for asterisk itself but for all the other bits one needs to have a usable home pbx setup? I'm mainly having trouble with terminoligy for the various hardware bits i expect to need... |
| 14:11.46 | [TK]D-Fender | ~book |
| 14:11.46 | 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 |
| 14:11.48 | [TK]D-Fender | ^^^^ |
| 14:12.20 | [TK]D-Fender | _AxS_, the words "home PBX" are typically an oxymoron. Residential homes don't normally have PBX's. |
| 14:12.39 | [TK]D-Fender | _AxS_, As for equipmement it depends what you intend to physically interface to. |
| 14:13.03 | lvlinux | my home does :-) |
| 14:13.04 | [TK]D-Fender | _AxS_, If you intend to use telco lines you'll need an interface to those. Then you'll need a PBX to use that with. |
| 14:13.05 | lvlinux | lol |
| 14:13.13 | [TK]D-Fender | _AxS_, Then probably phones. |
| 14:13.34 | [TK]D-Fender | _AxS_, As for the phones, those may require something else to connect to that PBX as well. All depending |
| 14:13.51 | [TK]D-Fender | _AxS_, So witth that in mind... tthere isn't much more "hardware" to think of. |
| 14:15.45 | _AxS_ | yeah -- phones (incl. softphones), some box to interface with phone lines, likely some box to interface with analog phones too but i'm not decided on that.. i'm just having trouble knowing what box-1 and box-2 are called; everything i search for tends to just say "voip" and might mention sip but thats about it. it's for home use so i'm not looking for professional-grade hardware here (mainly due to expected cost) |
| 14:16.25 | _AxS_ | An ATA is something that interfaces to analog phones right, not analog lines? or does it do both? |
| 14:16.39 | [TK]D-Fender | _AxS_, So you actually want to use analog phone lines? |
| 14:16.47 | [TK]D-Fender | ~ata |
| 14:16.47 | infobot | rumour has it, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 14:17.06 | [TK]D-Fender | ATA is typically FXS port: meant fo you to plug a phone into, not a line |
| 14:17.11 | _AxS_ | Eventually no, but i have them at the moment so for testing (and if hardware isn't too expensive) i figure yes. |
| 14:17.42 | [TK]D-Fender | _AxS_, if you're thinking "for testing" then I wouldn't bother. You don't need to use actual lines to test |
| 14:18.13 | [TK]D-Fender | You can use your phones to simulate the flow a caller would go through. This is better than wasting money on temporary hardware you won't use in production |
| 14:18.58 | [TK]D-Fender | You could do itt with 2 soft-phones straight up. Have the first one call the extension you'd have your inbound call from the outside land on and that will be what the outside caller would experience. |
| 14:19.26 | [TK]D-Fender | Learning only takes you having something to run * on, and preferrably 2 devices to use as "phones" (soft-phone, etc) |
| 14:19.44 | [TK]D-Fender | And you can get a full feel for how your system will work. menu call-flow, voicemail, etc |
| 14:20.03 | _AxS_ | yeah i've got a two-softphone setup already. ok i'll keep playing that way. |
| 14:20.46 | [TK]D-Fender | That'll tell you if you've got the flow working the way you want. Then you can consider what actual hardware you'd put money into |
| 14:22.38 | _AxS_ | On a separate note... i've noticed through googling that it seems gtalk / google-voice / whatever support is pretty much dead due to it dropping XMPP? Has there been any work or progress in developing a module that works with Google Hangouts? |
| 14:22.51 | [TK]D-Fender | nope |
| 14:23.00 | [TK]D-Fender | #nofreelunch |
| 14:23.14 | _AxS_ | too bad.. oh well. |
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| 14:29.37 | Brimstar | We've got an asterisk derived system and we're looking to add some cordless handsets, probably SIP-DECT since I've not had great luck with Wifi phones. Any equipment suggestions? |
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| 14:33.50 | [TK]D-Fender | Snom's as "OK" from what I hear, and better reports from a Europeon MFG whose name I'm having trouble recalling.. |
| 14:33.59 | [TK]D-Fender | They were a bigger telcom hardware maker... |
| 14:34.08 | [TK]D-Fender | but here's a site witha few few models : http://www.telephonydepot.com/Catalog/Wireless-IP-Phone |
| 14:35.18 | Brimstar | We've got a polycom system currently, but the range extenders are crap, system has to be rebooted regularly, and the interface makes doing much other than answering and talking a nightmare |
| 14:35.20 | nixnothing | morning |
| 14:35.25 | *** join/#asterisk rafaels (~rafaels@187.59.80.159) |
| 14:36.24 | Brimstar | and since we've added offices on the second floor of our current building they're not covered. Nor is the executive office area. Repeater drops to often for it to be stable |
| 14:39.21 | TandyUK | yealink w52's are good |
| 14:39.27 | TandyUK | and theyre range entenders work :) |
| 14:39.35 | TandyUK | their* |
| 14:53.07 | Brimstar | wow. The Yealink gear looks SOO much like the snom gear it isn't funny |
| 14:57.45 | Brimstar | Hm.. Yealink setup doesn't seem to ahve a multi-base station setup and with multiple floors that may not be a great fit. Talking to the people at the site linked. Seem to be thinking that 1 N700 per floor, 1 repeater to cover the extra area on the 1st floor, and some handsets will round it out pretty well |
| 14:59.13 | Brimstar | snom setup |
| 14:59.15 | *** join/#asterisk elyob (~elyob@host86-160-116-229.range86-160.btcentralplus.com) |
| 14:59.23 | jrun | what are certified/* tags? |
| 15:01.11 | jrun | i guess what i'm asking is what's the diff between certified/* tags and LTS [13.9.1] versions? |
| 15:02.40 | [TK]D-Fender | Unless you're getting paid support from Digium you're wasting your time with cert |
| 15:05.10 | *** join/#asterisk mjordan (mjordan@nat/digium/x-vhpmjwizawipzfez) |
| 15:05.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:11.16 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 15:14.25 | jrun | [TK]D-Fender: i see, thanks. |
| 15:15.09 | *** join/#asterisk areski (~areski@120.red-88-27-170.staticip.rima-tde.net) |
| 15:18.08 | *** join/#asterisk elyob (~elyob@host86-160-116-229.range86-160.btcentralplus.com) |
| 15:20.11 | _AxS_ | Does digium's switchvox iOS app work with plain asterisk or do you need to have their commercial package? |
| 15:21.35 | mjordan | You need Switchvox. |
| 15:22.49 | _AxS_ | k thx |
| 15:22.54 | _AxS_ | figured as much but wasn't certain |
| 15:24.01 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 15:25.11 | *** join/#asterisk nickgaw (nickg@SDF.ORG) |
| 15:25.39 | nickgaw | Hi, Where is a good place to find good quality asterisk techitions? |
| 15:27.40 | Penguin | looks around |
| 15:28.04 | Penguin | waits for the rest of the story |
| 15:28.55 | [TK]D-Fender | Here? |
| 15:28.57 | [TK]D-Fender | There? |
| 15:29.14 | [TK]D-Fender | #asterisk-consultants ? |
| 15:29.19 | nickgaw | I have looked around on line but not found much as most of them will ony deal with businesses? |
| 15:29.31 | [TK]D-Fender | State your needs |
| 15:29.39 | [TK]D-Fender | Can't advise until we know. |
| 15:30.59 | nickgaw | I am looking for a good person who knows how to properly setup asterisk either with or without a web interface and to get my extensions as well as callrecording working either where the calls are recorded one party in each channel of a stereo file or different files per caller. |
| 15:31.21 | nickgaw | I already have a sip company I ported my numbers to. |
| 15:31.50 | Penguin | Is there more to it? That sounds straight forward enough. |
| 15:32.56 | nickgaw | I am also trying to find good hardware based sip phones that either are wifi based or where they plug into the router and the handsets are chordless and someone to set them up for me so when I get them I just plug them into the router and they work? |
| 15:33.21 | [TK]D-Fender | Usually I'd just use an ATA with a DECT cordless phone |
| 15:33.48 | *** join/#asterisk mojtaba (~mojtaba@2620:101:f000:700:8d1c:bba2:4774:95a5) |
| 15:34.03 | [TK]D-Fender | I got some seriously cheap phones that work through my entire 60,000 sqft warehouse flawlessly for $40 plugged into an ata worth the same |
| 15:34.09 | nickgaw | If the phone has an answering machine in it can this work with asterisk for one of the lines being ported? |
| 15:34.18 | [TK]D-Fender | EW |
| 15:34.27 | [TK]D-Fender | If you're using the phone with * let * handle voicemail.... |
| 15:34.34 | Penguin | You can, but asterisk has voicemail. |
| 15:35.05 | lvlinux | _AxS_: Google did NOT drop support for XMPP or GVoice over XMPP. It still works w Asterisk and other software (such as Yate). |
| 15:35.11 | Penguin | But I have found out that old people would rather use the answering machine than have to retrieve voicemail |
| 15:35.12 | nickgaw | true some members of my family who will be using one of the lines don't know much about asterisk that is why I am asking about that. |
| 15:36.25 | _AxS_ | lvlinux: for me it doesn't matter either way, as i'm not located in the US. |
| 15:36.51 | Penguin | But the fact is, a person needs not know ANYTHING about asterisk to retrieve voicemail. You pick up the handset, press the Messages key if there is one or dial the voicemail extension if no button, and follow the instructions on which key to press for which action. Press 1 for new messages, etc. |
| 15:37.44 | nickgaw | What about the person to configure my asterisk setup properly and I am ok with using asterisk for voicemail? |
| 15:38.09 | nickgaw | I have written some people but none of them have responded. |
| 15:38.25 | Penguin | Is this a paid gig? |
| 15:39.55 | nickgaw | How much would someone want per hour to get this working? |
| 15:40.10 | _AxS_ | nickgaw: could you elaborate on what exactly it is you want working? |
| 15:40.14 | Penguin | Did you already install the OS? |
| 15:40.22 | _AxS_ | nickgaw: do you have asterisk installed already? |
| 15:40.37 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 15:40.41 | Penguin | Having asterisk installed already isn't very important. |
| 15:40.54 | Penguin | Having a working OS with adequate remote access is. |
| 15:41.08 | nickgaw | yes I was messing around with either using a debian asterisk setup or incrediblepbx with the FreePBX or another web interface. |
| 15:41.22 | _AxS_ | nickgaw: you have extensions set up? |
| 15:41.44 | _AxS_ | nickgaw: err, the phones i mean? |
| 15:42.10 | nickgaw | not yet. and no sip phones yet either as I was wanting to get asterisk working first with soft phones. |
| 15:42.10 | _AxS_ | (i'm using softphones so i did this in sip.conf) |
| 15:42.28 | _AxS_ | nickgaw: ok so you set something up in sip.conf and have the softphones working then. |
| 15:42.45 | Penguin | You'll set up hard phone in sip.conf too. |
| 15:43.18 | _AxS_ | nickgaw: what is the identifier you used in sip.conf for the phone(s) ? |
| 15:43.27 | Penguin | nickgaw: Are you wanting to learn anything about asterisk setup and administration or just have someone else roll out the system for you? |
| 15:43.40 | *** part/#asterisk Brimstar (~Brimstar@about/windows/staff/brimstar) |
| 15:44.11 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:6355:b61:14e7:dbba) |
| 15:44.47 | *** join/#asterisk swimmercol (~lancelot@4E9D6FC8.rev.sefiber.dk) |
| 15:44.54 | swimmercol | Hi guys |
| 15:45.02 | swimmercol | I am getting this message when try to make a call: |
| 15:45.19 | swimmercol | [2016-05-27 11:45:12] ERROR[4868][C-00000004]: rtp_engine.c:397 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? |
| 15:45.33 | swimmercol | when i try to load res_rtp_asterisk module |
| 15:45.35 | swimmercol | i got: |
| 15:45.42 | lvlinux | swimmercol: do you have an rtp.conf file? |
| 15:45.49 | swimmercol | libsrtp.so.0: cannot open shared object file: No such file or directory |
| 15:45.59 | Penguin | swimmercol: Did it ever work before? |
| 15:46.05 | swimmercol | yes |
| 15:46.11 | swimmercol | yes it work before |
| 15:46.12 | Penguin | And then you upgraded something? |
| 15:46.14 | swimmercol | it is Asterisk 13 |
| 15:46.27 | lvlinux | what did you do before it stopped working? |
| 15:46.35 | swimmercol | just reboot the server |
| 15:46.38 | swimmercol | and stop working |
| 15:46.39 | Penguin | Seems like you may need to recompile asterisk and reinstall it. |
| 15:46.54 | Penguin | Some things must have upgraded and left asterisk in an incompatible state. |
| 15:46.57 | lvlinux | yup |
| 15:47.09 | swimmercol | wow |
| 15:47.13 | swimmercol | maybe openssl? |
| 15:47.27 | *** join/#asterisk nickgaw (nickg@SDF.ORG) |
| 15:47.46 | Penguin | I wouldn't think so. |
| 15:47.50 | _AxS_ | swimmercol: lddtree /path/to/libsrtp.so.0 ..and whatever it says is missing, is what the problem will likely be |
| 15:48.08 | nickgaw | My client disconnected having the extensions setup is something I would like help with and yes I would like to learn how to manage things on my own just someone to teach me. |
| 15:48.36 | lvlinux | nickgaw: have you read the * book? |
| 15:48.38 | _AxS_ | nickgaw: ok -- sip.conf , what sections are set up in there other than [general] ? |
| 15:48.40 | Penguin | nickgaw: Teaching/learning takes much time. |
| 15:49.16 | lvlinux | nickgaw: the book gives you a full walkthrough of a basic setup and helps you understand everything. |
| 15:49.19 | lvlinux | ~book |
| 15:49.19 | 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 |
| 15:49.50 | nickgaw | the one at asteriskdocs.org? yes |
| 15:50.06 | swimmercol | is there a way to yum downgrade ? |
| 15:50.13 | Penguin | yes |
| 15:50.46 | Penguin | swimmercol: But it would be better to upgrade asterisk now rather than downgrade everything else that changed. |
| 15:50.53 | *** join/#asterisk nickgaw (nickg@SDF.ORG) |
| 15:51.07 | nickgaw | I have read the asteriskdocs book yes. |
| 15:51.15 | swimmercol | mmm |
| 15:51.18 | Penguin | ~book |
| 15:51.18 | 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 |
| 15:51.33 | Penguin | Heh, someone just did that a minute ago. |
| 15:52.11 | nickgaw | yes I have read that book. |
| 15:52.18 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-plkrehuobgrvktrl) |
| 15:53.38 | nickgaw | Why do you ask if I have read the book? |
| 15:54.13 | Penguin | Reading the book is usually the first recommended step to learning about the operation and administration of asterisk. |
| 15:54.52 | nickgaw | yes and there is a lot of information in that book. |
| 15:55.03 | Penguin | You're right about that. |
| 15:55.24 | Penguin | Do you have time to learn and then set up asterisk, or do you need asterisk done ASAP and then learn about it later? |
| 15:56.29 | _AxS_ | nickgaw: /msg me, i'll step you through getting your voicemail working so it doesn't pollute this channel so much. |
| 15:56.38 | nickgaw | I would like to get things working sooner then later but I am ok with learning as I don't wanto miss calls to the ported numbers as currently they are just going to our cell phones threw the sip company I used. |
| 15:58.32 | Penguin | The basic steps to getting phone calls are this: install asterisk, configure your ITSP settings and phones in sip.conf, configure dialplan (extensions) in extensions.conf, configure end-points to communicate with asterisk. |
| 15:58.39 | nickgaw | Do most asterisk technitions not support the GUI's? |
| 15:59.00 | Penguin | I will not support a GUI for asterisk. |
| 15:59.20 | Penguin | I admin via ssh. |
| 15:59.46 | _AxS_ | nickgaw: you specified GUIs or not. most techs are going to edit the .conf files directly since that's where the power is -- GUIs are for people that don't want to learn, mostly |
| 16:00.24 | nickgaw | ok understandable? |
| 16:00.25 | _AxS_ | my asterisk install didn't come with a gui; tbh it'd be more work to install it than it is to learn which .conf file to edit. |
| 16:01.03 | [TK]D-Fender | _AxS_, unless you use a pre-baked ISO |
| 16:01.13 | Penguin | There's really only a handful of confs to edit. |
| 16:01.25 | nickgaw | Then does it matter what hosting platform I use for hosting asterisk? |
| 16:01.36 | _AxS_ | [TK]D-Fender: yes -- that form of install would come with one; i shoved it on my linux server so just plain old asterisk for me |
| 16:01.40 | lvlinux | nickgaw: you want it to be reliable. Other than that you're good. |
| 16:02.07 | lvlinux | (I don't support any GUI as well, although I have made a couple very simple web interfaces.) |
| 16:02.08 | Penguin | nickgaw: I would recommend a distro with an LTS model. |
| 16:02.24 | lvlinux | second that ^^^^^^^^^^ |
| 16:02.26 | _AxS_ | nickgaw: and reliable in this case means way more reliable than most servers you likely have set up. |
| 16:02.51 | lvlinux | reliable for web hosting != reliable for VoIP |
| 16:02.58 | _AxS_ | nods in agreement |
| 16:03.01 | nickgaw | Is it ok to use a hosting company to host it for me? |
| 16:03.08 | Penguin | Depends. |
| 16:03.31 | _AxS_ | nickgaw: i would expect you would probably want to use a Hosted-PBX package if you're going that way, rather than your own asterisk install, no? |
| 16:03.34 | Penguin | What company did you have in mind? I ran a customer on Linode for a couple of years with no serious issues. |
| 16:03.51 | *** join/#asterisk newtonr (~newtonr@173-21-146-94.client.mchsi.com) |
| 16:03.51 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:04.12 | nickgaw | I was looking at digitalocean. |
| 16:04.20 | lvlinux | ehhh, idk |
| 16:04.54 | lvlinux | I have one on Iniz that has been very good. And Serverpoint ColossusCloud is great too. |
| 16:04.58 | *** join/#asterisk areski (~areski@80.174.128.25.dyn.user.ono.com) |
| 16:06.53 | nickgaw | I was looking at incrediblepbx but that includes the GUI and I am not sure how uppdated that is. for a hosted system do I need dahdi and other tools or just asterisk? |
| 16:07.06 | drmessano | Dont use incrediblepbx |
| 16:07.17 | Penguin | I strongly discourage crap like that. |
| 16:07.33 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 16:08.09 | Penguin | CentOS+asterisk, or Debian+asterisk, or any other regular stable distro +asterisk |
| 16:08.15 | nickgaw | my sip company sip.us has a module to set my trunks up that they say only works in freePBX so not sure if it could be done manually or not? |
| 16:08.16 | drmessano | Obviously this whole convo has been steering you away from a GUI, but if you want a GUI, thats not the distro to use |
| 16:08.39 | Penguin | Nothing only works in FreePBX. |
| 16:09.03 | drmessano | nickgaw: FreePBX just configures standard SIP peers for Asterisk... it all translates both ways |
| 16:09.16 | nickgaw | Should I talk with them first before deciding? |
| 16:09.26 | Penguin | Not really. |
| 16:09.30 | drmessano | No |
| 16:09.50 | Penguin | It comes down to if YOU need to have a GUI or if YOU can handle it all from the CLI. |
| 16:10.44 | drmessano | If you NEED a GUI, use the FreePBX distro.. the official one.. not some hack job fork. Thats all I will say before backing away from the G word, carefully and before anyone gets hurt |
| 16:10.56 | nickgaw | Are there combinations where I can use the GUI for some tasks but not for others like my advanced recording settings? |
| 16:11.05 | Penguin | no |
| 16:11.10 | Penguin | All GUI or no GUI. |
| 16:11.10 | drmessano | Yes |
| 16:11.15 | drmessano | Thats not true at all |
| 16:11.25 | _AxS_ | GUIs, do they manage the .conf files or do they read the configuration from a mysql db or some such? |
| 16:11.56 | [TK]D-Fender | They generate conf's based on their structures usually stored in DB's like mysql |
| 16:12.00 | Penguin | For a person just breaking into asterisk, I would never suggest hybrid GUI/no-GUI. |
| 16:12.15 | drmessano | You can create lots of custom dialplan with FreePBX, and override things it does in the GUI.. It doesnt read the config files.. it writes its own.. there are ways to add config in though, and override it |
| 16:12.18 | Penguin | For a well-experienced admin who understands exactly what the GUI does, maybe. |
| 16:13.06 | _AxS_ | FreePBX's gui likely provides most if not all of the advanced options anyways though, wouldn't it? |
| 16:13.07 | drmessano | Thats a lot different than "no" |
| 16:13.13 | nickgaw | for new users what do you suggest I start out with? |
| 16:13.41 | Penguin | CentOS, asterisk, ssh, and vim |
| 16:13.50 | drmessano | ... |
| 16:14.09 | Penguin | and the asterisk book. |
| 16:15.18 | _AxS_ | nickgaw: ... s/vim/any other text editor that'll work over ssh that you are comfortable with/ |
| 16:15.35 | drmessano | nickgaw: Plenty of folks go either way at the start.. depends on your comfort level |
| 16:15.56 | drmessano | But feel free to choose your distro of choice and whichever editor you choose |
| 16:16.19 | drmessano | or a GUI, for that matter.. it's about what you need and what you can do.. Asterisk is served up many different ways |
| 16:16.19 | nickgaw | Does it matter what hosting company I use then? |
| 16:16.32 | drmessano | DO is OK, just dont use the $5 option |
| 16:16.33 | drmessano | Ever |
| 16:16.41 | drmessano | Linode is my preference |
| 16:17.36 | Samot | Not to trash the Asterisk Book but if it's in reference to the website...ugh. |
| 16:17.39 | Samot | 1.6? |
| 16:17.55 | _AxS_ | 1.8 isn't it? |
| 16:18.00 | Penguin | The last book I looked at was geared toward 1.8. |
| 16:18.05 | *** join/#asterisk nickgaw (nickg@SDF.ORG) |
| 16:18.06 | Penguin | Been a while since I reviewed it. |
| 16:18.12 | Samot | Even still. |
| 16:18.25 | Penguin | You can also run asterisk at home as opposed to in the cloud. |
| 16:18.28 | nickgaw | If I don't go with a GUI then does it matter what host I use? |
| 16:18.33 | drmessano | No |
| 16:18.40 | drmessano | Just dont use a piece of crap |
| 16:18.55 | nickgaw | any companies to avoid? |
| 16:19.00 | drmessano | Asterisk is realtime audio, not web sites |
| 16:19.00 | Penguin | I've even seen a FreePBX install on a Linode VPS. |
| 16:19.08 | Penguin | FreePBX the web app, not FreePBX the distro. |
| 16:19.20 | _AxS_ | nickgaw: companies for what? hosting? the sip trunk? |
| 16:19.26 | drmessano | Ive seen plenty of them on Linode |
| 16:19.29 | nickgaw | hosting. |
| 16:19.34 | drmessano | Like I said |
| 16:19.42 | drmessano | DO is fine, jst dont use the $5 option |
| 16:19.51 | drmessano | I prefer Linode myself |
| 16:19.54 | Penguin | You don't have to put it on a VPS, though. |
| 16:19.57 | drmessano | On and NO AWS micro crap |
| 16:20.19 | Samot | Definitely not micro on AWS. |
| 16:20.22 | drmessano | Anything that cheap is going to be on an overloaded host |
| 16:20.31 | drmessano | and audio doesnt like that |
| 16:20.33 | Penguin | You can put it on your spare computer and throw it in the closet. |
| 16:20.34 | [TK]D-Fender | Book's 4th ed = * 11 |
| 16:20.45 | _AxS_ | nickgaw: most, probably. you want something that can guarantee 99.99999999995% uptime and in a way that isn't going to lose any context if they have to switch the hosting server (i've not even looked into asterisk vs clustering but i cna't guess its simple) |
| 16:20.51 | Samot | I was talking bout the website. |
| 16:21.53 | nickgaw | Should I use the centos rpm packages for asterisk? |
| 16:22.25 | _AxS_ | instead of building from scratch? probably yes. |
| 16:22.43 | Penguin | You could. I always recommend packages first, even if you have to roll your own. |
| 16:23.02 | _AxS_ | on linux you pretty much always want the package manager to handle your stuff. |
| 16:23.21 | Penguin | When the package doesn't exist, that's when you learn how to make RPMs. |
| 16:23.26 | _AxS_ | otherwise upgrading becomes hell later on. |
| 16:23.42 | Penguin | Source install = not recommended for anything (in my opinion) |
| 16:24.22 | _AxS_ | Yep. if you want to build stuff from source, use gentoo as your distro. |
| 16:24.27 | Penguin | The package manager does a better job managing packages than most people. |
| 16:24.29 | drmessano | LOL |
| 16:24.33 | Samot | Wut? |
| 16:24.42 | Samot | I install Asterisk from source all the time. |
| 16:24.52 | drmessano | Slackware is so much better |
| 16:25.13 | Penguin | You can build anything from source that you want, but _I_ always recommend rolling into a package instead of installing from the source. |
| 16:25.43 | Penguin | It's a few more steps, but it keeps package management sane. |
| 16:25.47 | nickgaw | slackware has good packaging building you just build into a tree and then install it. |
| 16:26.02 | _AxS_ | if the package manager doesn't handle the package, you run into issues like what swimmercol had earlier. |
| 16:26.34 | Penguin | If you can't be bothered to figure out how to make your own packages, use checkinstall. |
| 16:26.36 | nickgaw | What was that? |
| 16:26.56 | _AxS_ | nickgaw: swimmercol's issue? a module wouldn't work anymore because some other library got updated. |
| 16:27.20 | drmessano | Sorry, I was being facetious. Use whichever distro you LIKE and PREFER.. and if you want to install from source, thats fine too.. You have much more control that way, especially with something like Asterisk that has MANY different modules, and usually the packages go with a standard set |
| 16:28.15 | drmessano | The whole 'Wait a few months and the maintainer may build it to include X" is tiring and occurs often with Asterisk |
| 16:28.49 | drmessano | If youre banging out boxes with a standard, basic feature set, packages are fine |
| 16:29.12 | nickgaw | Other then the book where is a good place to find a good asterisk technitionto help me out? |
| 16:29.22 | drmessano | The wiki |
| 16:29.28 | _AxS_ | nickgaw: googling for random howtos, generally |
| 16:30.01 | nickgaw | So none of you do that for a living? |
| 16:30.50 | Penguin | Some of us do, but a single family asterisk deployment usually doesn't have much of a budget for paying an experienced sysadmin. |
| 16:31.08 | nickgaw | How much would they charge per hour? |
| 16:32.16 | Penguin | It varies by their location and skill level. I've seen it range anywhere from $15 to $250 per hour. |
| 16:32.25 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 16:32.34 | Penguin | I would be in the middle. |
| 16:32.38 | _AxS_ | most likely would sit at the $80 -> $250 level. pretty much the same as IT tech. |
| 16:32.48 | _AxS_ | s/tech/support/ |
| 16:33.28 | drmessano | I'll charge you $300 an hour, if you like.. Didnt know this was an RFP |
| 16:33.52 | _AxS_ | infobot: sweet, thanks for implementing regex's! |
| 16:33.54 | lvlinux | lol I'm at $50 if I work slow, $100 if fast |
| 16:34.14 | _AxS_ | lvlinux: you have clients that are OK with you working slow? :) |
| 16:34.18 | nickgaw | So if I was wanting to talk with one of you about helping me out how would I do that and would it help if I first got asterisk installed? |
| 16:35.03 | nickgaw | what do you mean by here? |
| 16:35.05 | drmessano | So youre not willing to do any of this yourself? |
| 16:35.12 | lvlinux | _AxS_: I normally do thing by the job, rather than hourly, so it's mostly irrelevant. |
| 16:35.19 | nickgaw | I am willing to do this on my own yes. |
| 16:35.24 | _AxS_ | nickgaw: i already told ya, I'll help you get voicemail working right now if you /msg me and can pastebin some of your .conf's -- this is just based on the stuff i played with last night; i'm not any sort of expert |
| 16:35.27 | drmessano | I mean, thats the point of community support |
| 16:35.36 | drmessano | If you just have questions, ask them on IRC |
| 16:36.02 | drmessano | If you want someone to do it for you, then thats to the point of paying someone |
| 16:36.23 | lvlinux | nickgaw: if you are willing to do it on your own then there are multiple people here that will help you with that. You'll have to do your due diligence and follow instructions, but there's lots of nice guys (and gals) here that are happy to help. |
| 16:36.31 | tonsofpcs | maybe I should just put together an RFP... |
| 16:37.06 | nickgaw | I am ok with paying someone to get me started with setting this up. |
| 16:39.00 | *** join/#asterisk nickgaw (nickg@SDF.ORG) |
| 16:39.31 | nickgaw | I am willing to pay someone to help me get started and to teach me how it is done so I can continue myself. |
| 16:40.22 | lvlinux | nickgaw: you have to make the decision if you want to pay or not. You can get to your goal either way :-) |
| 16:40.38 | _AxS_ | pays it forward -- helps others to learn after he's learned |
| 16:40.44 | drmessano | nickgaw: *Teaching* you is like 10 years work |
| 16:40.59 | drmessano | nickgaw: I suggest if you want to learn, follow whats being told in here |
| 16:41.09 | nickgaw | I am totally ok with paying someone to help me with this if they could also teach me how to do the rest on my own. |
| 16:41.15 | drmessano | nickgaw: If you just want someone to do it for you, make plan, spend the money |
| 16:41.34 | lvlinux | step #1 is read the book. It's free and it teaches you everything you need to know to get started. It will SAVE you time and effort. |
| 16:41.40 | drmessano | I doubt someone is going to take payment to teach you asterisk |
| 16:41.46 | lvlinux | I would :-) |
| 16:41.51 | lvlinux | but it ain't cheap |
| 16:42.04 | nickgaw | How much would you charge? |
| 16:42.05 | drmessano | No, not in the context of a couple hours work |
| 16:42.18 | drmessano | People pay thousands for Asterisk courses |
| 16:42.37 | drmessano | and even then.. that touches on concepts |
| 16:42.51 | drmessano | All you need is right in front of you |
| 16:43.13 | lvlinux | I do 1 on 1 mentoring but it's not worth it if someone has the patience to read and learn on their own with help from IRC, etc. |
| 16:43.59 | nickgaw | lvlinux can I message you my email address so we can talk off of irc? |
| 16:44.09 | lvlinux | nickgaw: sure if you like |
| 16:44.20 | _AxS_ | nickgaw: ...you know, if you asked some specific implementation questions we likely could've given you the info you needed to get working whatever it is you're working on, about 30 mins ago :) |
| 16:45.06 | lvlinux | _AxS_: yup :) |
| 16:45.10 | Penguin | If someone would have jumped on it, that would have been a billable 30 mins. |
| 16:45.14 | nickgaw | Would using nano rather then vim be ok for the default editor? |
| 16:45.19 | Penguin | YES |
| 16:45.19 | _AxS_ | nickgaw: yes |
| 16:45.20 | lvlinux | yes |
| 16:45.25 | Penguin | Any editor will suffice. |
| 16:45.25 | _AxS_ | any text editor you want. |
| 16:45.36 | _AxS_ | except emacs. |
| 16:45.38 | _AxS_ | :P |
| 16:45.41 | lvlinux | as long as it's a real text editor it's fine (i.e. no MS WORD) |
| 16:45.45 | lvlinux | emacs lol |
| 16:45.48 | Penguin | If you can use emacs, go for it. |
| 16:45.55 | drmessano | EDLIN is nice |
| 16:45.57 | lvlinux | who can use emacs though? lol |
| 16:46.03 | Penguin | precisely |
| 16:46.35 | lvlinux | you have to have 10 inch hands and exercise them daily to be able to. |
| 16:46.53 | lvlinux | all yo uhave to do with vim is remap CapsLock to ESC and you're good lol |
| 16:46.54 | nickgaw | I like nano better as I myself am totally bind and find someo of those GUI's tricky to use with any screen readers so editing the configuration files by hand would probably be a much more accessible way to set asterisk up right? |
| 16:47.14 | lvlinux | ah, yes absolutely |
| 16:47.42 | lvlinux | so the book may be a problem then |
| 16:48.17 | lvlinux | nickgaw I think you're the first person EVER that has been in this room that has a valid excuse for not jumping on the book immediately. :-) |
| 16:48.21 | nickgaw | I think that book has images in it with no alt texts |
| 16:48.25 | Penguin | Need a GUI? Use gvim instead of vim. :D |
| 16:49.27 | nickgaw | Is there an on line version of the 4th addition I can read on line or just the pdf of an earlier addition? |
| 16:50.22 | _AxS_ | nickgaw: you can likely buy it as epub -- that should be readable via screen-reader |
| 16:50.50 | nickgaw | yes epub has accessibility built into the format. |
| 16:51.45 | _AxS_ | although the 4th edition is available as free PDF too and it seems mostly text based; don't know if that would work for you or not? |
| 16:52.27 | nickgaw | yes it would work just fine is that on the asteriskdocs web site? |
| 16:53.16 | _AxS_ | nickgaw: it was posted earlier: http://asterisk-service.com/downloads/Asterisk-%20The%20Definitive%20Guide,%204th%20Edition.pdf |
| 16:54.05 | nickgaw | What version of asterisk does it cover? |
| 16:54.12 | lvlinux | version 11 |
| 16:54.44 | lvlinux | almost everything is applicable to version 13, except PJSIP in v13, which is optional to use. |
| 16:55.00 | _AxS_ | ... how advanced has asterisk's webrtc support come along since version 11 ? |
| 16:55.09 | lvlinux | v11 = nonexistant |
| 16:55.12 | lvlinux | v13 = working |
| 16:55.31 | lvlinux | webrtc support was introduced in 12 |
| 16:58.06 | nickgaw | so to download that pdf ebook using wget would I just put the entire URL in quotes? |
| 16:58.11 | lvlinux | yes |
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| 16:58.38 | _AxS_ | shouldn't even need quotes, i think. |
| 16:58.51 | lvlinux | probably not but it won't hurt |
| 16:58.55 | _AxS_ | nods |
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| 17:01.08 | fish9370 | hello |
| 17:01.12 | lvlinux | hello |
| 17:01.20 | fish9370 | help me |
| 17:01.20 | swimmercol | omg |
| 17:01.33 | fish9370 | I have some problem with CDR |
| 17:01.41 | swimmercol | Rackspace just updated some kernel packages by ansible :S |
| 17:01.46 | swimmercol | just fixed |
| 17:01.49 | swimmercol | lol |
| 17:05.18 | fish9370 | who knows Asterisk? |
| 17:05.41 | [TK]D-Fender | Probably EVERYONE here |
| 17:05.50 | [TK]D-Fender | You know where you are... right? |
| 17:05.53 | [TK]D-Fender | this is #asterisk |
| 17:06.20 | fish9370 | why I can't replace my own variable on channel? |
| 17:06.47 | Penguin | Not all variables are writable. |
| 17:06.53 | fish9370 | before call I set CDR(trunk)='mytrunk' |
| 17:07.00 | [TK]D-Fender | what variable? |
| 17:07.06 | [TK]D-Fender | Yuo need to SHOW US the failure |
| 17:07.15 | fish9370 | variable about trunk |
| 17:07.20 | [TK]D-Fender | We have no idea what you're talking about, how you're doing anything, or what the real result is |
| 17:07.33 | [TK]D-Fender | <fish9370> variable about trunk <- these words do not mean anything |
| 17:07.52 | [TK]D-Fender | Please give a PROPER description and SHOW US the failure |
| 17:07.54 | [TK]D-Fender | ~pb |
| 17:07.54 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:07.55 | [TK]D-Fender | ^^^^^ |
| 17:08.49 | fish9370 | please look at my log http://pastebin.com/JUafRP9U |
| 17:10.59 | [TK]D-Fender | Where in there? |
| 17:11.13 | [TK]D-Fender | What line shows the thing you're having a problem with? |
| 17:12.20 | fish9370 | I have this result in CDR http://pastebin.com/5wGvvTQn |
| 17:13.11 | fish9370 | I try raplace variable calltype _before_ call, but it finalized |
| 17:13.26 | [TK]D-Fender | Where do we see this? |
| 17:16.12 | fish9370 | 5 sec, I update my log for you |
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| 17:22.10 | fish9370 | there is now log http://pastebin.com/QYvB3646 |
| 17:22.51 | fish9370 | look at line 27 and line 50 |
| 17:23.35 | fish9370 | first I set calltype = IN and after calltype = TRANSIT |
| 17:24.35 | fish9370 | and how you can see in CDR (there is in bottom of log) calltype in all cdr records is IN |
| 17:25.19 | _AxS_ | OT: does debian/other distros have 'wgetpaste' available? |
| 17:27.37 | fish9370 | it's because CDR created _after_ Dial and I create it on channel _before_. How I can cheat this? |
| 17:27.41 | [TK]D-Fender | I don't see full channel names anywhere so I can't match those up. |
| 17:28.01 | [TK]D-Fender | Also you should have AGI debug enabled for this |
| 17:28.09 | [TK]D-Fender | So we can prove exactly how things were called |
| 17:29.10 | fish9370 | sorry for my English, you undestand me? |
| 17:29.33 | [TK]D-Fender | Show us the call with AGI debug enabled so we can prove what you're setting and how |
| 17:29.59 | fish9370 | ok, I anable AGI debug, CDR debug |
| 17:31.09 | fish9370 | there is http://pastebin.com/E8vhF6su |
| 17:38.26 | fish9370 | look at line 101 (set variable to IN) and 183 (set variable to TRANSIT) |
| 17:44.26 | [TK]D-Fender | [May 27 20:30:29] WARNING[17765]: cdr.c:2957 ast_cdr_setvar: channel 1: SIP/0001034-00000003, calltype: IN |
| 17:44.35 | [TK]D-Fender | but no line like this following it like the others had |
| 17:45.09 | fish9370 | what you mean? |
| 17:46.11 | fish9370 | oh, it's my own comment in source code |
| 17:47.15 | fish9370 | I need set variable after Dial and before Dial end |
| 17:47.24 | fish9370 | how I can do this? |
| 17:48.16 | [TK]D-Fender | on the CALLED channel? |
| 17:50.39 | fish9370 | yes |
| 17:50.54 | [TK]D-Fender | you should do that using a pre-dial hook. |
| 17:50.59 | [TK]D-Fender | This is supported in * 11+. |
| 17:51.06 | [TK]D-Fender | please refer to the WIKI for this |
| 17:51.17 | fish9370 | where I can read about this? |
| 17:51.34 | fish9370 | on 13+ it's possible? |
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| 18:04.13 | [TK]D-Fender | [TK]D-Fender> This is supported in * 11+. |
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| 19:12.41 | F2Knight | Having a core dump issue - not sure what is causing it any help would be wonderful - Backtrace located here > https://gist.github.com/tuxpowered/6a8ecc6037548aed24a7f51500bc3e25 |
| 19:16.17 | _AxS_ | signal 6.. havnen't seen that one before. |
| 19:22.43 | F2Knight | _AxS_ is that sarcasi |
| 19:22.50 | F2Knight | sarcasm ? * |
| 19:23.23 | _AxS_ | F2Knight: nope -- i haven't had a core dump with signal 6 that i can remember. tons of signal-11's , lots of other random signals too. |
| 19:24.12 | F2Knight | asterisk-dev suggests that its an issue with the mysql-connector and odbc and to upgrade them but sadly they are not available on FreePBX. |
| 19:25.30 | _AxS_ | #6 is the Abort signal, so that's something at runtime that is expressly aborting the code run.. you could grep for any 'signal' calls that send SIGABRT to find which one it is...? |
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| 19:27.21 | F2Knight | I honestly do not know how to do that. |
| 19:27.53 | _AxS_ | that said, looking at the backtrace, odbc is disconnecting and libodbc is trying to free stuff when the abort occurrs... if i had to guess i would agree that the issue is likely in libmyodbc |
| 19:28.05 | F2Knight | I mean if you are talking just 'grep' the back trace sure... its the single line stating the terminated |
| 19:28.16 | _AxS_ | no i'm talking grep of the code. |
| 19:28.40 | _AxS_ | the original source, likely libmyodbc |
| 19:28.50 | F2Knight | oh .. yeah I wouldn't know what to do if I did see that. Asterisk-dev channel suggested updating ODBC and Mysql-connector. |
| 19:29.30 | F2Knight | just no updates available. |
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| 19:29.56 | _AxS_ | ... was glibc recently upgraded? the SIGABRT seems to follow a 'malloc_printerr' but I see no such error .. or is there some other stderr or debug output available in a log? |
| 19:31.53 | F2Knight | No this is a clean install of FreePBX distro |
| 19:31.56 | _AxS_ | is asterisk multithreaded or multiprocess? |
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| 19:32.08 | F2Knight | how can i tell? |
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| 19:37.04 | _AxS_ | F2Knight: that was more a question for the channel... multi-threaded generally means memory's shared, while multi-process means each concurrent piece has its own memory segment. if libmyodbc is freeing memory allocated in a different segment, then i could see why it's causing this. |
| 19:37.16 | _AxS_ | ..that said i don't think code like that would work nearly as well as it has so far |
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| 20:52.05 | lvlinux | whats the difference in BUSY and INUSE device states? |
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| 21:14.14 | rubio | Hi! can anyone recommend a good SIP world wide provider with API for account management? |
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| 21:35.42 | swimmercol | try twilio |
| 21:39.05 | rubio | i see on previous logs that someone also recommend flowroute... |
| 21:39.52 | rubio | @swimmercol do you have any opinion choosing between them? |
| 21:40.14 | swimmercol | not really |
| 21:40.23 | rubio | ok |
| 21:44.59 | lvlinux | rubio: do be aware that Twilio bills by the minute, while Flowroute and some others are 6 seconds. |
| 21:48.09 | rubio | lvlinux: interesting data! flowroute price looks more expensive (for longs calls)... also, i'm not sure but twilio looks to have less coverture? |
| 21:48.28 | lvlinux | rubio: AnveoDirect is another for you. |
| 21:48.38 | lvlinux | coverture? |
| 21:48.57 | rubio | coverage, sorry... |
| 21:49.00 | lvlinux | ah |
| 21:49.23 | lvlinux | Twilio SIP is .0045 I think. Anveo Direct is .004. Flowroute is .01. |
| 21:50.43 | rubio | i can't find the prices for channels and DIDs... :( |
| 21:52.29 | lvlinux | Twilio is I think $1/mo, Flowroute is somewhere around $1.50, AnveoDirect is $0.19/mo |
| 21:52.33 | lvlinux | for US DIDs that is. |
| 21:53.49 | lvlinux | Twilio doesn't do incoming channels I don't think. It's unlimited number, all per minute. Flowroute is also unlimited channels, but they have a virtual PRI with one or two for I think $25. |
| 21:54.09 | rubio | looks like flowroute is the most expensive and anveoDirect the cheapest... |
| 21:54.34 | lvlinux | yes among those three, but Flowroute isn't expensive. 1c/min is pretty standard. |
| 21:54.45 | rubio | have you try them? do you have any opinion on voice quality? |
| 21:55.12 | lvlinux | I haven't used Twilio, but voice quality on AnveoDirect and Flowroute for me has been excellent. No issues at all. |
| 21:55.42 | lvlinux | DIDlogic is another, but I'm not sure they have an API for account management. They're $.007/min |
| 21:56.30 | lvlinux | They're also a good provider, excellent voice quality, worldwide POPs, etc. |
| 22:02.33 | rubio | as you said, DIDlogic does not looks to provide api.. :( |
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| 22:48.14 | rubio | it looks like flowroute does not provide international dids... :( |
| 23:21.29 | lvlinux | rubio: ah, I didn't know that. Well AnveoDirect and Twilio do. |
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