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| 01:38.51 | Samot | lankanmon: if ypu want to uss POTS youre looking at needing a FXO card. |
| 01:39.49 | Samot | Or something like a SPA3102 or whatever the new model is called. |
| 01:41.17 | lankanmon | wow those are expensive! Are those the only way to get the pc to connect to POTS |
| 01:41.20 | lankanmon | ? |
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| 01:41.59 | WIMPy | You use a card or some VOIP gateway. |
| 01:42.17 | *** part/#asterisk gtjoseph (~circuser-@unaffiliated/gtj) |
| 01:42.49 | lankanmon | are there any budget friendly gatways that will work with asterisk? |
| 01:43.49 | WIMPy | One has just been named. |
| 01:44.03 | lankanmon | SPA3102? |
| 01:44.25 | WIMPy | yes |
| 01:45.11 | lankanmon | Is this it? https://www.amazon.ca/Cisco-Voip-Single-Gateway-spa3102-na/dp/B007I57DCS/ref=sr_1_2?s=electronics&ie=UTF8&qid=1457574262&sr=1-2&keywords=SPA3102 |
| 01:45.42 | lankanmon | looks like $122 is the lowest price |
| 01:46.04 | WIMPy | Any model with an FXO port. |
| 01:47.46 | lankanmon | I am not sure that I know the difference between that and normal VOIP modems... I have a OBi202, would that work? |
| 01:48.12 | Samot | Does it have an FXO port? |
| 01:48.40 | WIMPy | What is a VOIP modem? Doesn't make sense. |
| 01:48.57 | lankanmon | Im not sure what to call it |
| 01:49.05 | lankanmon | OBi202 VoIP Phone Adapter with Router |
| 01:49.11 | lankanmon | https://www.amazon.ca/OBi202-Phone-Adapter-Router-2-Phone/dp/B007D930YO/ref=sr_1_2?s=electronics&ie=UTF8&qid=1457574436&sr=1-2&keywords=obihai |
| 01:49.39 | babak | Hi , I installed latest dahdi on ubuntu 12.04 and 14.04 , I checked header file version=ok ,but every time I should run modprobe dahdi, "make config" on dahdi compile supposed to do this ,am I correct ? |
| 01:51.16 | WIMPy | cannot parse that. |
| 01:52.17 | Samot | lankanmon: you need something woth an FXO port not just FXS. |
| 01:52.37 | lankanmon | oh there is a difference |
| 01:52.45 | Samot | FXO = in. FXS = out |
| 01:53.20 | Samot | Fax machine, analog phone...all FXO devices |
| 01:53.47 | Samot | The wall jack you connect them to, FXS |
| 01:53.56 | Samot | Soif |
| 01:54.31 | Samot | So if you want to take your phone company POTS line and have tye PBX answer it, it needs to accept FXO connections. |
| 01:54.32 | lankanmon | So I need one with both ports right? |
| 01:54.58 | Samot | Well now youre looking at two devices. |
| 01:55.07 | Samot | You have two phones. |
| 01:55.35 | Samot | You either need a single device with at least one FXO port and 2 FXS portsm |
| 01:55.47 | lankanmon | No, Just one phone, one line in and possibly one voip connection |
| 01:55.47 | Samot | And thoe arent cheap. |
| 01:56.31 | Samot | So a 3102, which is EOL |
| 01:57.06 | Samot | Or the 232D which is not EOL or any other brand |
| 01:58.43 | lankanmon | This? https://www.amazon.ca/Cisco-SPA232D-G1-Multi-Line-DECT/dp/B00BBM7RJS/ref=sr_1_1?ie=UTF8&qid=1457575085&sr=8-1&keywords=spa232d |
| 01:59.03 | Samot | Yeah |
| 01:59.34 | Samot | That will support up to 5 phones if you get the proper DECT base. |
| 02:00.04 | lankanmon | lets say I get this, how do I apply this with Asterisk |
| 02:00.13 | WIMPy | That's built-in. isn't it? |
| 02:00.20 | Samot | Yup |
| 02:00.35 | lankanmon | No Idea... I did not know about asterisk until yesterday |
| 02:00.48 | lankanmon | I am trying to learn as much as possible |
| 02:00.51 | WIMPy | Not a bad idea. |
| 02:01.06 | Samot | You need to read the manual on how to config the FXO port and bridge to the PBX |
| 02:01.20 | WIMPy | Ugh. It's really only DECT ;-( |
| 02:01.24 | lankanmon | is the PBX running on a different system? |
| 02:01.34 | Samot | Yes. |
| 02:01.41 | Samot | You need a system for that |
| 02:01.46 | lankanmon | oh |
| 02:01.54 | Samot | These are external devices. |
| 02:02.08 | Samot | You need to build a linux box |
| 02:02.15 | lankanmon | yep |
| 02:02.22 | Samot | Install Asterisk and any other things you need. |
| 02:02.38 | lankanmon | would it work with a Rasp Pi 3? |
| 02:02.54 | lankanmon | or 2? |
| 02:02.55 | Samot | Yes |
| 02:03.03 | WIMPy | HA. COMPLETELY FORGOT ABOUT THAT. |
| 02:03.08 | WIMPy | oops |
| 02:03.47 | lankanmon | Its just that i really dont want to build a big system that I will run 24hrs/day for just phone service |
| 02:04.32 | Samot | It's really never wise to run other things on a PBX but for 1 line, that shouldn't really be a problem. |
| 02:05.01 | lankanmon | yeah its purely for personal use |
| 02:05.57 | lankanmon | How would I connect the adapter to the rasp? |
| 02:06.18 | WIMPy | Ethernet |
| 02:06.34 | lankanmon | oh |
| 02:06.43 | lankanmon | I thaught usb or something |
| 02:07.20 | WIMPy | It's called Vo*IP* for a reason :-) |
| 02:07.28 | Samot | You wouldn't connect the adapter to the PBX |
| 02:07.44 | Samot | You would connect it to the router on the LAN the PBX is on. |
| 02:08.34 | lankanmon | And asterisk will be able to find that? |
| 02:08.42 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 02:08.44 | Samot | No. |
| 02:08.56 | Samot | The ATA will tell Asterisk that it wants to connect. |
| 02:09.09 | Samot | And there will be an extension/user on Asterisk with a username and password that you setup. |
| 02:09.16 | Samot | You put that in the ATA |
| 02:09.18 | WIMPy | You have to configure the device to find Asterisk. |
| 02:09.27 | lankanmon | oh |
| 02:09.38 | Samot | ~book |
| 02:09.39 | 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 |
| 02:09.43 | Samot | There you go. |
| 02:10.11 | Samot | ~wiki |
| 02:10.24 | Samot | http://wiki.asterisk.org |
| 02:10.27 | lankanmon | so: Line in -> SPA232D -> router -> Rasp Pi -> voip service ? |
| 02:11.04 | Samot | No. |
| 02:11.35 | lankanmon | which part is wrong? |
| 02:11.37 | Samot | POTS to SPA232D to Router to Pi |
| 02:11.57 | Samot | VOIP service to Pi via SIP trunk. |
| 02:12.12 | Samot | Pi to router to SPA232D to phone |
| 02:12.45 | Samot | It doesn't matter if the call originates from the POTS or the VOIP service. |
| 02:12.58 | Samot | When the call hits Asterisk, you tell it what to do. |
| 02:13.59 | lankanmon | Oh, so I will be able to direct calls that come from POTS strait to voip service, but calls from voip service need to be validated by pin and then it must ask for a number to dial... |
| 02:14.07 | lankanmon | is that do able? |
| 02:15.35 | lankanmon | (like a call service - but for myself) - that is why it needs pin to validate that it is me |
| 02:16.01 | WIMPy | Whatever you configure. |
| 02:16.56 | lankanmon | ok |
| 02:17.10 | lankanmon | for now, I got AsteriskNow working in a VM |
| 02:17.26 | lankanmon | I will play around with it while I learn and get the hardware together |
| 02:20.34 | Samot | What do you mean by VOIP service? |
| 02:21.23 | lankanmon | one sode connecta to pots other to voip. so i can call though voip to a local number |
| 02:25.52 | Samot | So there is no VOIP provider. |
| 02:26.03 | Samot | No one that gives you a connection to/from the PSTN |
| 02:26.12 | Samot | That's only done on the POTS line? |
| 02:27.00 | lankanmon | i have voip service. i want to connect to it |
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| 02:29.02 | Samot | So why would you route an incoming call from the POTS to the VOIP provider? |
| 02:29.13 | Samot | Unless you're CFing to your cell phone. |
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| 02:36.36 | Pegasus_RPG | Hello. Anyone here experienced with Snom phones, specifically the 300? I'm just trying to activate TLS again after a firmware update and the phone is saying the certificate issuer is not recognized. It's issued by an intermediate CA whose cert I uploaded to the phone, and the parent CA is already present. |
| 02:36.36 | Pegasus_RPG | Sure I can add an exception but I shouldn't have to |
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| 02:42.38 | lankanmon | Samot: So the thing is I live in canada, the rest of my family are in asia |
| 02:42.50 | lankanmon | it costs a lot to communicate with them via phone |
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| 02:43.21 | lankanmon | so last year I went and setup a voip service at my home there so I can talk to my family |
| 02:43.40 | lankanmon | but now I want to expant that so I can also make outgoing calls to local numbers. |
| 02:43.44 | Samot | Like I said, if you're doing for that you're good. |
| 02:43.57 | lankanmon | that is why I hope to setup this system |
| 02:43.59 | Samot | But you only need to do that if you're remote. |
| 02:44.16 | lankanmon | yes I will be in canada, so I need a reliable system |
| 02:44.17 | Samot | Otherwise you would just dial directly out of the VOIP provider. |
| 02:44.41 | lankanmon | yes my phone service in canada is fully voip |
| 02:45.22 | lankanmon | the call rates for local calls there is cheap compared to the rates I pay to call from here |
| 02:45.33 | lankanmon | so thats why I want to set it up |
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| 03:11.35 | lankanmon | Samot - Does the VOIP service connect to asterisk through SOA232D or through the PI (asterisk running on PI) |
| 03:12.40 | Samot | Well it would depend on your service with them and what they allow |
| 03:12.58 | Samot | Sounds like you have a typical service which they expect to be on an ATA |
| 03:13.09 | lankanmon | yes i think so |
| 03:13.15 | Samot | You're going to want to get a SIP Trunk. |
| 03:13.16 | lankanmon | i just connect it to a phone |
| 03:13.30 | Samot | You're going to need to talk to them about it |
| 03:13.34 | Samot | What they allow, etc. |
| 03:14.23 | lankanmon | would i be able to use it with my current by connecting my VOIP device to the SOA232D as a line? |
| 03:18.02 | lankanmon | Can u see this? drathir |
| 03:18.29 | lankanmon | https://awwapp.com/b/ubegzoadd/ |
| 03:18.43 | lankanmon | I drew out what I have in my head |
| 03:23.10 | Samot | You would have to see what the 232D can do. |
| 03:23.22 | Samot | But technically you could but your VOIP account on the FXS line. |
| 03:24.25 | lankanmon | but then don't i need 2 ports? |
| 03:28.52 | lankanmon | How does this look? https://www.amazon.ca/Grandstream-GS-HT702-Handytone-Telephone-Adapter/dp/B007PEIHKE?ie=UTF8&*Version*=1&*entries*=0 |
| 03:29.03 | lankanmon | its cheaper... which would be better for me |
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| 03:37.29 | lankanmon | Samot: Also do younthink a rasp pi 2 model b can handle the tasks? I can't find a pi 3 for cheap ATM |
| 03:37.42 | Samot | Should be fine. You're not doing much with it |
| 03:38.18 | lankanmon | any thaughts on the grandstram HT702? |
| 03:38.31 | Samot | lankanmon: I personally despise Grandstream. |
| 03:38.31 | lankanmon | it is supposed to have 2 FXS ports |
| 03:39.02 | Samot | OK let's break this down. |
| 03:39.07 | Samot | Let's start with the PBX |
| 03:39.14 | Samot | PBX = Pi2 |
| 03:39.28 | lankanmon | yes |
| 03:39.33 | lankanmon | i get that |
| 03:39.36 | Samot | PBX will have one extension, your home phone. |
| 03:40.01 | Samot | In order to use that extension you need an ATA or a Softphone. |
| 03:40.14 | lankanmon | ok |
| 03:40.18 | Samot | Now you want to receive and make calls |
| 03:40.19 | lankanmon | I have ATA |
| 03:40.54 | Samot | You have both POTS and VOIP service. |
| 03:41.01 | lankanmon | yes |
| 03:41.10 | Samot | Now do you want both to connect to the PBX? |
| 03:41.45 | lankanmon | I want to place this setup in the other side of the world, and call it via VPN |
| 03:42.01 | Samot | Let's not get a head. |
| 03:42.07 | lankanmon | It will ask me for the number I want to dial and then forward |
| 03:42.31 | Samot | Where is the PBX going to be physically located? |
| 03:42.46 | lankanmon | in another country |
| 03:43.05 | Samot | OK and how will it be able to send and receive calls? |
| 03:43.10 | Samot | Will it have a VOIP provider? |
| 03:43.14 | lankanmon | yes |
| 03:43.29 | Samot | OK |
| 03:43.30 | lankanmon | I currently have a voip setup there but I can only call that one home |
| 03:43.45 | lankanmon | I want to call that home + other local homes at a local rate |
| 03:43.48 | Samot | So now you want to be able to call into your home and forward that call out? |
| 03:44.01 | lankanmon | yes |
| 03:44.08 | lankanmon | percisely |
| 03:44.24 | Samot | Why does the PBX need to be in another country? |
| 03:44.50 | lankanmon | to handle calls and dial out |
| 03:45.00 | lankanmon | I am not sure how to keep it here |
| 03:45.04 | Samot | It doesn't physically need to be in the country to do that. |
| 03:45.08 | Samot | It's an IPBX |
| 03:45.14 | Samot | Internet. |
| 03:45.21 | lankanmon | Hmmmm... |
| 03:45.28 | Samot | Here's what you do |
| 03:45.33 | Samot | Setup the PBX in your house. |
| 03:45.39 | Samot | Create an extension for you. |
| 03:45.51 | Samot | Create an extension for your mom (example) who's in Asia. |
| 03:46.13 | Samot | Now your mom just needs an ATA that's programmed to connect on her extension to the PBX |
| 03:46.33 | lankanmon | yes that is my current setup. |
| 03:46.44 | Samot | So if you're extension 1000 and she's 1001, you can just pick up the phone and dial 1001 |
| 03:47.07 | Samot | Then you're just calling a "local" extension on the PBX |
| 03:47.11 | lankanmon | but lets say I want to call my aunt... or uncle.. So I want to be able to call theior landline so as to avaoid giveing everyont an atx |
| 03:47.15 | lankanmon | *ATA |
| 03:47.17 | Samot | $0 billing. |
| 03:47.23 | Samot | OK |
| 03:47.37 | lankanmon | yes I currently dial 102 and it calls there for free |
| 03:47.49 | lankanmon | I just want to call out as well... |
| 03:48.07 | Samot | Remotely? |
| 03:48.49 | Samot | Locally? Both? |
| 03:49.01 | Samot | So like you're at the mall and want to call mom.. |
| 03:49.07 | Samot | You want to call home and then call her? |
| 03:49.48 | lankanmon | no |
| 03:49.59 | Samot | So you just want to do this from home? |
| 03:50.22 | lankanmon | I call the extension on my phone and call to asia (my home there fore free) |
| 03:50.38 | Samot | OK |
| 03:50.41 | lankanmon | I then want PBX to ask me for a local number. when I dial, it wil connect me |
| 03:50.50 | Samot | Local? |
| 03:50.53 | Samot | To who? |
| 03:50.54 | lankanmon | I can then talk to my aunt |
| 03:51.01 | lankanmon | local to asia |
| 03:51.07 | lankanmon | long distance from here |
| 03:51.12 | Samot | No, International. |
| 03:51.19 | Samot | Long Distance is Yukon |
| 03:51.24 | lankanmon | yes |
| 03:51.40 | lankanmon | So the rates are super high otherwise |
| 03:51.42 | Samot | Who is your VOIP provider? |
| 03:51.49 | lankanmon | but local rates are tiny |
| 03:51.53 | lankanmon | VOIP.ms |
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| 03:53.31 | Samot | They provide the ATA or is it yours? |
| 03:53.43 | lankanmon | it is mine on both sides |
| 03:53.56 | lankanmon | one is an Obihi202 |
| 03:54.00 | Samot | So you really don't need a PBX to make this work. |
| 03:54.03 | lankanmon | the other is a grand stream |
| 03:54.06 | Samot | You already have it working. |
| 03:54.16 | lankanmon | but I can not dial out |
| 03:54.20 | Samot | Why not? |
| 03:54.21 | lankanmon | it is a closed system |
| 03:54.34 | Samot | They haven't activated International on your account probably. |
| 03:54.56 | Samot | Doesn't matter if you have a PBX or not. If they won't send International calls, nothing is going to work. |
| 03:55.05 | lankanmon | oh, no I can call international, but the rates are still much higher than local |
| 03:55.29 | lankanmon | I currently have it set as an extention |
| 03:55.32 | Samot | What do you mean? |
| 03:55.38 | Samot | Local = Local dialing to you |
| 03:56.19 | lankanmon | I can make local calls on my end and the device in asia can also make local calls in my end, but I can not make local calls there... |
| 03:56.31 | Samot | Right. |
| 03:56.43 | Samot | Because the account is setup based on your location. |
| 03:56.49 | lankanmon | I want to make and recive local calls there |
| 03:56.53 | lankanmon | yes |
| 03:57.07 | Samot | OK, you need to either have VOIP.ms turn on International dialing for you |
| 03:57.27 | Samot | Or, if you put the PBX there, you need a provider there that will handle the calls. |
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| 03:57.49 | lankanmon | but I can make international calls from voip.ms, but the rates are super high |
| 03:57.49 | Samot | It doesn't matter if I take my IP phone to the UK. |
| 03:58.01 | Samot | Then you need to find a carrier with lower rates. |
| 03:58.18 | Samot | It's based on your number. |
| 03:58.27 | lankanmon | wouldn't PBX negate that totally? |
| 03:58.32 | Samot | No. |
| 03:58.57 | lankanmon | because I will be dialing an extention (free) then calling a local number there (local rates vs international) |
| 03:59.03 | Samot | You still need to connect the PBX to a carrier that will handle the calls to/from the PSTN |
| 03:59.07 | Samot | No. |
| 03:59.25 | Samot | Your extension in Asia is an extension of your Canadian based account. |
| 03:59.40 | Samot | It's why they can call Canada with standard rates. |
| 03:59.49 | Samot | If I take my IP phone to the UK... |
| 03:59.59 | Samot | All my calls to the US/Canada are going to be my normal rates. |
| 04:00.09 | Samot | But if I call a number in the UK, it's still International. |
| 04:00.21 | Samot | Because my number is a US number. |
| 04:00.55 | lankanmon | can't pbx act like a phone card service whare it asks for a ew number and dials it? |
| 04:00.57 | Samot | You can get a International number from VOIP.ms in that area. |
| 04:01.02 | Samot | Sure. |
| 04:01.14 | Samot | But it still needs to send the call to a carrier/provider. |
| 04:01.20 | Samot | Who then routes that call over the PSTN |
| 04:01.34 | Samot | There is still a charge. |
| 04:01.37 | lankanmon | yes I will be getting a second line to my home there |
| 04:01.54 | lankanmon | that way my relatives can call me on that number and it will ring over here in canada |
| 04:02.21 | Samot | OK. So if you get a number in Asia.. |
| 04:02.29 | Samot | Send your calls out that number. |
| 04:03.03 | lankanmon | yes |
| 04:03.20 | Samot | But the PBX can still physically be at your house. |
| 04:03.47 | Samot | You could even go to someplace like Vultr or Digital Ocean and get a VM server. |
| 04:03.47 | lankanmon | and connect to the telephone adapter through the net? |
| 04:03.55 | Samot | Yes, that's the point of it. |
| 04:04.05 | lankanmon | ok that sounds doable |
| 04:04.14 | lankanmon | I just hope there is not much lag |
| 04:04.23 | jfindley | That works out pretty well. |
| 04:04.45 | jfindley | I just took on a client in India doing the same thing. There's a little latency, but the audio quality is good. |
| 04:04.50 | Samot | Well no matter where you put the PBX there will be latency for one side. |
| 04:05.03 | Samot | It's about 300-400ms there. |
| 04:05.23 | Samot | I've had customer all over India. |
| 04:05.25 | lankanmon | the regular calls I make work well, if I can get that, then I am happy |
| 04:05.51 | Samot | But basically you need an International DID |
| 04:06.07 | Samot | Or look at other providers like flowroute.com |
| 04:06.57 | lankanmon | they do not sell international DIDs |
| 04:07.06 | Samot | The only way you're going to have $0 billing between you and Asia is if all the end points are attached to the PBX. |
| 04:07.21 | Samot | https://voip.ms/intldids.php |
| 04:07.24 | Samot | They sure do. |
| 04:07.59 | lankanmon | I am not looking for $0 billing, just cheap billing on the asia side and free connection between |
| 04:08.52 | Samot | What country? |
| 04:09.10 | lankanmon | Sri Lanka |
| 04:09.41 | lankanmon | it is an uncommon country, so phone companies do not offer good rates |
| 04:09.48 | Samot | Nope. |
| 04:10.23 | Samot | I think it's actually consider High Cost which limits access. |
| 04:11.15 | lankanmon | probably |
| 04:11.16 | Samot | What is the cost per minute for you now? |
| 04:11.30 | lankanmon | 13 cents/m |
| 04:12.16 | Samot | That's standard. |
| 04:12.51 | Samot | If you look really hard you might find someone that could have it cheaper. |
| 04:12.52 | lankanmon | yeah, but local rates will be around 2 cents/min |
| 04:13.02 | Samot | But it's not going to be much cheaper. |
| 04:13.14 | Samot | Again, get a Sri Lanka based DID. |
| 04:13.22 | lankanmon | not sure how |
| 04:13.57 | lankanmon | their tech is a bit behind and phone companies are not too happy about VOIP |
| 04:15.19 | Samot | Most phone companies have adopted VOIP |
| 04:16.41 | lankanmon | not them... they are still trying to milk the international charges |
| 04:18.15 | Samot | VOIP.ms? |
| 04:18.46 | Samot | VOIP.ms is not a phone company. |
| 04:18.55 | Samot | They are a VOIP provider. |
| 04:19.08 | lankanmon | i was talking about the phone companies in SL |
| 04:19.32 | Samot | Well unfortunately, you're going to be stuck with about $0.13/minute |
| 04:19.40 | lankanmon | voip.ms does not have SL dids |
| 04:19.45 | Samot | Nope. |
| 04:19.47 | Samot | Not many do. |
| 04:20.00 | lankanmon | yeah thats why i want to rool my own system |
| 04:20.05 | Samot | It's not a country North American countries generally play nice with. |
| 04:20.18 | lankanmon | I was really happy with the one I setup last year |
| 04:20.23 | Samot | Unless you're going to roll out your own backbone and interconnects with carriers... |
| 04:20.38 | Samot | You still need a carrier/provider. |
| 04:20.43 | lankanmon | cant i jus tconnect to normal phone line |
| 04:20.52 | Samot | POTS? |
| 04:20.53 | Samot | Sure. |
| 04:20.55 | lankanmon | yes |
| 04:20.59 | lankanmon | that was my plan |
| 04:21.03 | Samot | And .13/minute will double. |
| 04:21.09 | lankanmon | why? |
| 04:21.22 | Samot | Because generally those carriers are still using TDM. |
| 04:21.51 | lankanmon | not firmiliar with TDM |
| 04:21.56 | Samot | Copper wires. |
| 04:22.12 | Samot | Traditional phone service. |
| 04:22.16 | lankanmon | so, its not possible to connet to POTS? |
| 04:22.23 | Samot | Yes it is. |
| 04:22.32 | lankanmon | I thaut that is why you need FXS |
| 04:22.39 | Samot | Yes. |
| 04:22.43 | lankanmon | instaed of FXO |
| 04:22.46 | Samot | No. |
| 04:22.54 | Samot | FXS = out |
| 04:23.08 | Samot | The phone jack in your wall is FXS |
| 04:23.14 | Samot | The jack on your ATA is FXS. |
| 04:23.26 | Samot | The jack on your analog phone is FXO |
| 04:24.01 | lankanmon | So, why can it not be connected to the wall over there? |
| 04:24.13 | Samot | Because the call still has to go through someone. |
| 04:24.24 | Samot | There is infrastructure involved here. |
| 04:24.40 | Samot | That connection has to get from you to the other side. |
| 04:24.46 | lankanmon | will it not go though the sri lankan phone provider? |
| 04:24.48 | Samot | That's what the provider does. |
| 04:25.00 | Samot | Sure, if you want to deal with them |
| 04:25.08 | Samot | And have to setup over there locally. |
| 04:25.24 | lankanmon | I was just planning on setting up a second line there |
| 04:25.35 | lankanmon | I will be going there this summer and can set it up |
| 04:25.56 | Samot | Are there VOIP providers there? |
| 04:25.59 | Samot | That offer SIP trunks? |
| 04:26.09 | lankanmon | none that i am aware of |
| 04:26.25 | lankanmon | the market is dominated by 3 big companies and it is hard to penetrate |
| 04:26.36 | lankanmon | lots of financial motivation |
| 04:27.04 | Samot | The only other way you can do this is two PBX systems. |
| 04:27.16 | Samot | One there and one with you |
| 04:27.29 | lankanmon | how will that work? |
| 04:27.31 | Samot | Asia PBX connects via FXS/FXO to the local carrier. |
| 04:27.46 | Samot | Canada PBX connects via SIP to VOIP.ms |
| 04:28.03 | Samot | And you can use a SIP or IAX2 trunk to connect the two PBXes together |
| 04:28.41 | lankanmon | but why cant the PBX server in SL just connect straight to voip.ms from there? |
| 04:28.58 | lankanmon | then I just connect to it as an extention |
| 04:28.59 | Samot | Sure. |
| 04:29.25 | Samot | At that point I would just put a PBX in the cloud. |
| 04:29.53 | Samot | Oh but you can't because you need FXS/FXO |
| 04:29.56 | lankanmon | maybe I will, I need to do a cost benefit anaysis to see what option is cheaper |
| 04:30.07 | Samot | You can put the PBX in SL |
| 04:30.10 | lankanmon | true |
| 04:30.26 | Samot | You'll still need the FXS/FXO card or device. |
| 04:30.39 | lankanmon | https://www.amazon.ca/gp/product/B007PEIHKE/ |
| 04:30.44 | lankanmon | I was thionking of this one |
| 04:30.50 | lankanmon | it is relatively cheap |
| 04:30.56 | Samot | No |
| 04:31.01 | Samot | You need something with FXO |
| 04:31.13 | Samot | You need something that you can plug the phone line from the SL carrier into. |
| 04:31.21 | Samot | FXS is OUT |
| 04:31.22 | lankanmon | oh |
| 04:31.37 | Samot | In the layout. |
| 04:32.05 | lankanmon | Ok I will need to look around |
| 04:32.19 | lankanmon | Will you be around tomorrow? |
| 04:32.39 | lankanmon | I have to go now... |
| 04:32.47 | lankanmon | I really appriciate all your help |
| 04:33.03 | Samot | I'm always online, it will be a case if I see the message. |
| 04:33.41 | lankanmon | ok thats fine... I am usuall always online as well, just AFK |
| 04:37.39 | jfindley | What's the best way to track a call through the dialplan? e.g., a call comes in to some context, a dial is made which executes a macro when the call is answered, then back to the context for cleanup. Currently I use a channel variable set with __ in the beginning, but is there a better way? |
| 04:38.31 | jfindley | err, at each of these points I make AGI calls where logic dependent on the call metadata takes place |
| 04:38.57 | jfindley | something like agi_threadid would be nice, but that ID changes when the agi call is made in the macro. |
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| 13:47.36 | pckz | hey |
| 13:48.35 | pckz | we have a bug with asterisk, in the automatic menu with phone numbers. All devices works perfect, but only iphones got freeze when the first key we press is "1" key |
| 13:48.45 | pckz | is just in iphones |
| 13:48.55 | pckz | and only with key 1 |
| 13:49.08 | pckz | after that doesn't works any other number of the menu |
| 13:49.49 | [TK]D-Fender | then that's a problem on the iphone, not an Asterisk problem. |
| 13:50.09 | [TK]D-Fender | Asterisk can't make an iphone "freeze" |
| 13:50.22 | pckz | we've tested with a lot of devices |
| 13:50.28 | pckz | nop sorry |
| 13:50.34 | pckz | isn't freeze |
| 13:50.52 | pckz | is just asterisk continue the automatic response |
| 13:51.02 | pckz | but can't get any option |
| 13:51.51 | [TK]D-Fender | Show us the actual call |
| 13:51.55 | [TK]D-Fender | ~pb |
| 13:51.55 | 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. |
| 13:51.57 | [TK]D-Fender | ^^^ |
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| 13:53.49 | pckz | haha sorry |
| 13:54.07 | pckz | how can I show you the call |
| 13:55.42 | [TK]D-Fender | * CLI |
| 13:59.04 | pckz | actually doesn't works with any key |
| 13:59.33 | pckz | humm what does * CLI means?. Sorry for this, first time that I see asterisk. But I learn fast hahaha :( |
| 14:00.01 | [TK]D-Fender | asterisk -rvvvvvvvvvvvvvvvvvvv |
| 14:00.09 | [TK]D-Fender | If you don't know CLI then you haven't learned your basics |
| 14:00.11 | [TK]D-Fender | ~book |
| 14:00.12 | 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 |
| 14:00.13 | [TK]D-Fender | ^ |
| 14:00.47 | [TK]D-Fender | And I find it hard to believe you'd ahve gone to the point of actually creating an IVR yourself without having gknown how to use * CLI |
| 14:01.28 | pckz | nop, I'm try to fix |
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| 14:03.21 | [TK]D-Fender | What do you mean "nop"? |
| 14:03.30 | [TK]D-Fender | YOU didn't configure your system? |
| 14:03.30 | pckz | but I didn't set up by myself |
| 14:03.35 | [TK]D-Fender | Who did? |
| 14:04.55 | pckz | I'm watching CLI and just reach : Probation passed - setting RTP source address to .... |
| 14:06.11 | [TK]D-Fender | That is impossible if you issued the command I gave you and you placed a call your systtem actually processed |
| 14:06.21 | pckz | and after that with my iphone ,asterisk can't detect the keys. But with android works like a charm :/ |
| 14:06.29 | [TK]D-Fender | Show the call |
| 14:08.17 | pckz | show the call means that I have to show you what is displayed by the CLI right? |
| 14:09.57 | [TK]D-Fender | yes |
| 14:10.00 | [TK]D-Fender | Go to CLI |
| 14:10.02 | [TK]D-Fender | place the call |
| 14:10.08 | [TK]D-Fender | PASTEBIN the whole thing |
| 14:10.11 | [TK]D-Fender | ~pb |
| 14:10.11 | 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. |
| 14:10.13 | [TK]D-Fender | ^^^ |
| 14:10.34 | [TK]D-Fender | IF you see NOTHING then your call isn;t even hitting your server and our time is being wasted |
| 14:11.49 | pckz | this is a call from iphone: http://pastebin.com/KWAHxUuE |
| 14:12.25 | pckz | and this is a call from an android: http://pastebin.com/d6aEVGqf |
| 14:12.30 | pckz | just works the android call :D |
| 14:13.32 | [TK]D-Fender | that is a call from your PROVIDER |
| 14:13.58 | [TK]D-Fender | qwhich means your DTMF mode isn't set right in your peer or you are using an AUDIO stream that is not stable enough |
| 14:14.08 | [TK]D-Fender | SIP/sipentel |
| 14:14.10 | pckz | sorry for the worng link with the iphone call, this is: http://pastebin.com/8b5FFv4d |
| 14:14.25 | [TK]D-Fender | ^ What "dtmfmode=" setting are you using for this in sip.conf? |
| 14:15.01 | pckz | dtmfmode=rfc2833 |
| 14:15.16 | [TK]D-Fender | pckzwe have a bug with asterisk, in the automatic menu with phone numbers. All devices works perfect, but only iphones got freeze when the first key we press is "1" key <- doesn't match this description yet. I don't see that iphone call getting ANY digit in first THEN failing |
| 14:15.47 | pckz | yep, that is the issue |
| 14:15.57 | [TK]D-Fender | try "inband" instead of "rfc2833" and test |
| 14:16.08 | [TK]D-Fender | make the change then do "sip reload" |
| 14:16.13 | [TK]D-Fender | from CLI |
| 14:16.22 | [TK]D-Fender | you should see it telling you that it is reloading your configs |
| 14:16.37 | WIMPy | Oh, BTW: Can anyone confirm that dtmfmode=inband works for sending DTMF for them? |
| 14:16.38 | pckz | thanks, I will try |
| 14:19.00 | [TK]D-Fender | WIMPy: Does if the carrier isn't filtering it. |
| 14:19.12 | [TK]D-Fender | WIMPy: Some do all sorts of stupid things |
| 14:19.21 | [TK]D-Fender | and... we're about to see... |
| 14:19.35 | WIMPy | It already sounds shreddered if I record the channel with MixMonitor. |
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| 14:23.23 | pckz | ohhh |
| 14:23.28 | pckz | thanks [TK]D-Fender |
| 14:23.45 | pckz | it's working! [anakin gif] |
| 14:24.23 | pckz | you're a god |
| 14:26.56 | [TK]D-Fender | No, far from, but I have a clue. |
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| 16:13.27 | jamesc | I don't see to have the REPLACE Function when was this introduced |
| 16:22.31 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:22.31 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.2 (2016/02/05), 11.21.2 (2016/02/11); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 16:22.32 | [TK]D-Fender | in any version 11+ |
| 16:36.33 | *** join/#asterisk busymind (~textual@c-24-21-79-235.hsd1.wa.comcast.net) |
| 16:36.35 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
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| 16:45.51 | jamesc | [TK]D-Fender: http://www.voip-info.org/wiki/view/Asterisk+func+REPLACE |
| 16:46.05 | [TK]D-Fender | That wiki is shit |
| 16:46.15 | [TK]D-Fender | not "the shit". The negative kind |
| 16:46.23 | jamesc | [TK]D-Fender: Is ther an alternative method. |
| 16:46.36 | [TK]D-Fender | the official WIKI. The source where you can see actual modules. |
| 16:46.40 | [TK]D-Fender | "core show functions", etc |
| 16:46.52 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_STRREPLACE |
| 16:46.59 | [TK]D-Fender | Which would show you this actual function |
| 16:47.24 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 16:47.57 | jamesc | [TK]D-Fender: is there a method for 1.6 compataible |
| 16:48.11 | [TK]D-Fender | go look at your function list |
| 16:48.24 | [TK]D-Fender | 1.6 is not supported, neither is 1.8, 10, or 12. |
| 16:48.48 | [TK]D-Fender | 11 Shouldn't even be considered since 13 is LTS and 11 ges into sec fix only in a few months. |
| 16:52.04 | file | time flies |
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| 17:04.01 | *** join/#asterisk retentiveboy (~retentive@2601:cf:4400:d8e4:785d:e061:42b0:b896) |
| 17:05.23 | retentiveboy | Does pjsip support getting external_*_address periodically from a DNS lookup to support setups behind a NAT'ing router with a dynamic public address? |
| 17:12.31 | [TK]D-Fender | nope |
| 17:12.40 | [TK]D-Fender | that's one of the key unfortunate things about it |
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| 17:21.18 | retentiveboy | crap |
| 17:22.25 | [TK]D-Fender | you could always script config file regens based on a polling script, but it's uglier than it should be. |
| 17:22.59 | retentiveboy | Yeah, ugly but sounds like it'll be required. RetentiveBoy isn't happy. |
| 17:24.28 | retentiveboy | related... is there syntax to include another file into one of the .conf files? |
| 17:25.22 | retentiveboy | (I should google first...) |
| 17:26.38 | *** join/#asterisk ledoktre (~Adium@216.51.224.229) |
| 17:26.53 | ledoktre | I was going to update to Asterisk Certified LTS 13.1-cert4 on Debian Jessie, and I'm running into a couple of quirks. Hoping someone here might be able to point me in the right direction. 1. configure script stops with a syntax error on line 19170 of configure script - near unexpected token ILBC and bombs. Anyone seen this? 2. pjsip is now apparently the defacto. What libraries do I need to satisfy the install? |
| 17:30.37 | [TK]D-Fender | TThere is no "defacto" |
| 17:30.42 | [TK]D-Fender | You use whatever you configure |
| 17:30.49 | [TK]D-Fender | chan_sip wasn't removed |
| 17:31.03 | [TK]D-Fender | also Cert = worthless unless you're paying for Digium support. |
| 17:31.16 | [TK]D-Fender | decimal revisions matter as they added real features between them |
| 17:35.42 | ledoktre | [TK]D-Fender. Yeah, Im not using premium support. Guess certified there means something a bit different than other places. Curious now - you are recommending just installing the regular 13.7.2 as opposed to any LTS or Certified version, right? |
| 17:37.31 | Samot | 13 is LTS |
| 17:38.29 | [TK]D-Fender | http://www.asterisk.org/downloads/asterisk/all-asterisk-versions |
| 17:38.34 | ledoktre | Yeah guess your right. I misread that. I was looking for LTS and saw it listed as 13.1.4 and thought it happened to be the same as Certified. My bad :-p |
| 17:38.35 | Samot | You just want 13 not Cert if no Digium support like [TK]D-Fender said |
| 17:39.16 | ledoktre | [TK]D-Fender: Yeah, Im on that page right now. I just mis-read it. Point well drilled, thanks Fender and Samot both. |
| 17:39.43 | ledoktre | I'll try compile with the non cert version see if the configure error persists |
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| 17:56.41 | retentiveboy | Is it safe to "core reload" when there are active calls? |
| 17:56.57 | retentiveboy | Well, "safe" as in won't drop calls. |
| 17:57.12 | [TK]D-Fender | nope. |
| 17:57.18 | [TK]D-Fender | wait... |
| 17:57.23 | [TK]D-Fender | sorry, that should be ok |
| 17:57.28 | retentiveboy | Same with "pjsip reload"? |
| 17:57.35 | [TK]D-Fender | reload just does the configs not a full module reload |
| 17:57.42 | [TK]D-Fender | should be safe |
| 17:58.44 | retentiveboy | the "pjsip reload" alias looks like "module reload res_pjsip*.so" under the hood |
| 17:59.29 | ledoktre | [TK]D-Fender - 13.7.2 resolved compile issue. I guess I completely misunderstood what "certified" meant. Thanks for straightening me out. |
| 17:59.45 | [TK]D-Fender | You're welcome |
| 18:01.44 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 18:06.09 | retentiveboy | [TK]D-Fender, "sorcery_object_load: Type 'transport' is not reloadable" |
| 18:06.29 | retentiveboy | Looks like reloading from a cron script is going to be really messy |
| 18:06.55 | retentiveboy | or is that unrelated? |
| 18:11.16 | *** join/#asterisk stephan48 (stephan@opennic/stephan) |
| 18:12.19 | [TK]D-Fender | That looks VERY related... and it's noticing a transport. |
| 18:12.49 | [TK]D-Fender | because the transport changes it thinks it might including changing a port binding even though our target is the advertised address |
| 18:12.56 | [TK]D-Fender | Might not actually be doable... which would suck |
| 18:13.08 | retentiveboy | Testing |
| 18:26.56 | retentiveboy | [TK]D-Fender, confirmed. "pjsip show transport transport-udp-nat" shows external_* values don't change after changing pjsip.conf and a "core reload". Crap. |
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| 18:43.24 | *** join/#asterisk dan_j (sid21651@gateway/web/irccloud.com/x-eljpaoedpnvayyqq) |
| 18:44.55 | dan_j | Hi. When using Queues and RingAll, is it possible to delay some agents from ringing? A client wants junior staff to ring first, and then add senior staff after 20 seconds if junior staff havent answered. |
| 18:45.31 | dan_j | My thought was if there was some sort of delay or penalty that can be set to a number of seconds. |
| 18:45.41 | dan_j | on a per agent basis. |
| 18:46.45 | [TK]D-Fender | There is |
| 18:46.48 | [TK]D-Fender | Read the tooltips |
| 18:47.10 | dan_j | I saw penalty but is that used when using ringall? |
| 18:47.30 | [TK]D-Fender | yes |
| 18:47.51 | [TK]D-Fender | Who is in a given level isn't tied to a specific strategy. |
| 18:48.02 | [TK]D-Fender | Where do you see "only works with type X"? |
| 18:49.46 | dan_j | I just assumed RINGALL does just that. |
| 18:49.48 | dan_j | "entries with higher penalties are considered last" |
| 18:49.57 | dan_j | What is there to consider if you are ringing all? |
| 18:51.20 | [TK]D-Fender | Who you're rining? |
| 19:08.13 | dan_j | If you are ringingall, then how does the penalty work? IE what effect does it have if it rings all anyway? |
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| 19:24.40 | dan_j | I think i've got it. I'm going to make two queues. The first only having junior members, and the second having junior and senior members. Calls will move from the 1st to the 2nd queue after 20 seconds if they arent answered, and the 2nd queue will have a higher weight so that it takes priority over the 1st queue. |
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| 19:43.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 19:45.08 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 21:04.52 | *** join/#asterisk panga (~panga@190.210.64.49) |
| 21:06.29 | panga | Hello |
| 21:06.45 | panga | Have someone receive this error before? |
| 21:06.47 | panga | <PROTECTED> |
| 21:06.47 | panga | <PROTECTED> |
| 21:06.47 | panga | <PROTECTED> |
| 21:06.47 | panga | <PROTECTED> |
| 21:06.50 | panga | <PROTECTED> |
| 21:06.53 | panga | <PROTECTED> |
| 21:06.56 | panga | <PROTECTED> |
| 21:06.58 | panga | <PROTECTED> |
| 21:07.00 | panga | <PROTECTED> |
| 21:07.04 | [TK]D-Fender | PASETBIN |
| 21:07.07 | panga | <PROTECTED> |
| 21:07.10 | [TK]D-Fender | DO NOT FLOOD IN HERE |
| 21:07.12 | panga | <PROTECTED> |
| 21:07.14 | [TK]D-Fender | ~pb |
| 21:07.15 | 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. |
| 21:07.16 | [TK]D-Fender | ^^^ |
| 21:07.17 | panga | [2016-03-10 17:54:41] ERROR[56484]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("dynamic", "(null)", ...): Name or service not known |
| 21:07.20 | panga | [2016-03-10 17:54:41] WARNING[56484]: acl.c:833 resolve_first: Unable to lookup 'dynamic' |
| 21:07.22 | panga | Sure |
| 21:07.41 | panga | here is |
| 21:07.42 | panga | http://pastebin.com/UThr9D7e |
| 21:08.01 | panga | I can call between phones |
| 21:08.09 | panga | in the same network |
| 21:08.19 | panga | but I can't call outside |
| 21:08.47 | [TK]D-Fender | <PROTECTED> |
| 21:08.49 | [TK]D-Fender | <PROTECTED> |
| 21:08.51 | [TK]D-Fender | Autofalltthough |
| 21:08.57 | [TK]D-Fender | very clearly a dialplan failure |
| 21:09.26 | panga | I'm using Macro instead of GoSub |
| 21:09.36 | [TK]D-Fender | that is a GOTO |
| 21:09.40 | panga | Since I'm migrating from and old version |
| 21:09.42 | [TK]D-Fender | And it failed |
| 21:10.57 | panga | So I have to recheck extensions.conf |
| 21:11.11 | [TK]D-Fender | You can see right there what it's doing |
| 21:13.06 | *** join/#asterisk CeBe (~CeBe@2a02:560:41dd:8400:5dbf:c663:cf79:65e8) |
| 21:13.18 | panga | No sure what I should check :-/ |
| 21:15.58 | *** join/#asterisk mjordan (mjordan@nat/digium/x-ldtonarkwdycnhoz) |
| 21:15.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:16.49 | panga | [TK]D-Fender: Do you have another clue to me? |
| 21:16.57 | [TK]D-Fender | Clue? |
| 21:17.01 | [TK]D-Fender | Thre is no clue. |
| 21:17.09 | [TK]D-Fender | You SEE the line number that is doing that Goto. |
| 21:17.11 | [TK]D-Fender | GO THERE |
| 21:17.15 | [TK]D-Fender | make your changes |
| 21:17.32 | [TK]D-Fender | <PROTECTED> |
| 21:17.33 | [TK]D-Fender | <PROTECTED> |
| 21:17.35 | [TK]D-Fender | <PROTECTED> |
| 21:17.36 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 21:18.12 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 21:19.45 | [TK]D-Fender | Executing [s@macro-trunkdial-failover-0.3:2] <----- |
| 21:29.58 | [TK]D-Fender | packs up and heads home |
| 21:32.25 | *** join/#asterisk CeBe (~CeBe@2a02:560:41dd:9200:456f:f851:5d70:abcf) |
| 21:40.00 | panga | I can receive incoming calls, but I can do outgoing calls, someone has that issue? |
| 21:40.38 | WIMPy | Is there a "not" missing? |
| 21:41.10 | panga | Fist time configuring asterisk :-/ |
| 21:44.40 | acidfoo_ | "The latest versions of the 79xx firmware finally support UDP communications" Jason Rose. https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 21:44.56 | acidfoo_ | does anyone know if its true that the latest firmware support SIP UDP ? |
| 21:49.42 | *** join/#asterisk ruied (~ruied@bl7-221-53.dsl.telepac.pt) |
| 21:52.13 | vader- | Can anyone recommend some fax settings? So I followed this guide to setup the Cisco SPA112 for faxing: https://www.t38fax.com/support/knowledgebase/cisco-spa112-2-port-phone-adapter/ |
| 21:52.37 | vader- | and I enabled T38 passthrough, as well as using a SIP provider that does T38 pass through |
| 21:52.40 | vader- | Flowroute |
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| 22:23.17 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 22:25.15 | [TK]D-Fender | panga |
| 22:25.15 | [TK]D-Fender | I can receive incoming calls, but I can do outgoing calls, someone has that issue? |
| 22:25.30 | [TK]D-Fender | Lots of people have had lots of issues. You'll have to actually look at yours |
| 22:25.43 | [TK]D-Fender | And properly state which one is NOT working... |
| 22:34.36 | *** join/#asterisk BlackMaria (~BlackMari@dsl.198.58.150.55.ebox.ca) |
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| 22:42.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:49.08 | lankanmon | Will this work with asterisk running on a rasp pi? https://www.amazon.ca/LINKSYS-SPA-3000-UNLOCKED-Analog-Adapter/dp/B00MFCMNNI/ |
| 22:49.15 | lankanmon | I need to connect to POTS |
| 22:49.25 | lankanmon | and to voip on the other ends |
| 22:49.48 | WIMPy | It will work with anythign that has a network connection. |
| 22:50.07 | lankanmon | so this should work right? |
| 22:50.18 | lankanmon | I am really on a budget and this would be great |
| 22:59.26 | lankanmon | Samot: You on? |
| 22:59.44 | Samot | About to eat dinner |
| 22:59.55 | Samot | Bsck in a while |
| 23:00.02 | lankanmon | ok |
| 23:00.15 | lankanmon | just take a look at the link when u have a chance... |
| 23:16.09 | rrittgarn | having an issue with the email notifications in voicemail.conf - specifically i am trying to print the ${UNIQUEID} in the email and it is just coming up blank - like the variable has no value. I have another server with almost identical voicemail.conf settings and no problems. Is there somewhere else that this feature is dependent on that i'm missing? |
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| 23:44.43 | Samot | lankanmon: its fine. Out of production for years but fine. |
| 23:45.08 | Samot | Ill be around later |
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| 23:51.07 | RadicalDev | With FastAGI, what's the best way to test if either the caller, callee, or both have hung up? |
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