| 00:00.01 | jpsharp | "hey siri, dial 202 456 1111 on speaker" |
| 00:02.08 | tuxinatorlinux | Is there a variable that indicates the status of a SIP peer (e.g. UNKNOWN, OK) |
| 00:03.33 | Samot | jpsharp: I didn't realize that IP desk phones have Siri. |
| 00:08.27 | Samot | And tuxd00d's theory also involves a key being pressed. The statement was about doing this without pressing keys... |
| 00:08.49 | tuxinatorlinux | Can something with HINTS do this? |
| 00:09.24 | tuxd00d | I don't think [J]oules wants to clarify. |
| 00:10.20 | Samot | tuxinatorlinux: You can use hints on Asterisk for the phones to subscriber to via BLF. |
| 00:10.33 | Samot | That will let you see the status of the device. |
| 00:11.07 | tuxinatorlinux | I'm wanting to have a dialplan action happen should one extension be unavailabe. |
| 00:11.50 | tuxinatorlinux | But before a particular extension is dialed. |
| 00:13.18 | tuxinatorlinux | For example, if the Internet is out at a location, all of the phones would be offline. I'd like a message that played to incoming calls in that situtaion. |
| 00:14.31 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints |
| 00:14.57 | tuxinatorlinux | I see that, but I don't see how to get the status from the dialplan. |
| 00:19.58 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 00:19.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0-rc1 (2018/07/03), Standard: 15.5.0-rc1 (2018/07/03); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:19.59 | tuxinatorlinux | That works. Thank you. |
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| 01:24.07 | tuxinatorlinux | What if I want to check more than one extension? Like: |
| 01:24.08 | tuxinatorlinux | GotoIf($["${EXTENSION_STATE(100@hints)}" = "UNAVAILABLE"] && $["${EXTENSION_STATE(101@hints)}" = "UNAVAILABLE"]?offline:online) |
| 01:31.03 | tuxinatorlinux | If one extension is 'UNAVAILABLE' and other is 'NOT_INUSE', it still routes to 'offline'. I'd like it to only route to offline if both extensions are 'UNAVAILABLE'. |
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| 03:50.32 | [TK]D-Fender | tuxinatorlinux, you are using && outside of an expression. |
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| 08:55.58 | wtf911 | i have a super easy question ;) |
| 08:56.27 | wtf911 | i'm compiling asterisk from the github master branch. which opus module file should i use? 64bit |
| 08:56.46 | wtf911 | or perhaps i should ask, which version of asterisk is the master branch? version 15? |
| 09:00.38 | file | master is what 16 will become |
| 09:00.43 | file | the 15 branch is 15, the 13 branch is 13 |
| 09:00.57 | wtf911 | ah file you're from the gvsip thread :D |
| 09:01.15 | wtf911 | i'm trying to try opus with google voice but am not sure which codec_opus.so to use |
| 09:01.39 | file | the 15 one would probably work |
| 09:01.58 | file | but dunnno for sure haven't tried it |
| 09:02.40 | wtf911 | [Jul 12 08:52:18] WARNING[3443]: loader.c:888 load_dlopen: Error loading module 'codec_opus.so': /usr/lib/asterisk/modules/codec_opus.so: undefined symbol: ast_sorcery_unref |
| 09:03.50 | file | ah someone changed it |
| 09:04.18 | file | then there isn't currently a compatible codec_opus, it'll exist when 16 occurs |
| 09:05.00 | wtf911 | :( |
| 09:08.16 | wtf911 | is there someone i can poke to put one out early? |
| 09:08.31 | wtf911 | http://downloads.digium.com/pub/telephony/codec_opus/ |
| 09:08.37 | wtf911 | perhaps into the unsupported folder |
| 09:10.30 | file | 16 doesn't exist yet, so it'd have to be a one-off one... you could email Matt Fredrickson, creslin@digium.com, but there's no guarantee he'd do such a thing |
| 09:11.36 | file | you could also just apply naf's change to Asterisk 13 and use codec_opus as normal there? |
| 09:15.17 | wtf911 | i believe you are correct it would work with 13 but to be honest i wouldn't know how to go about applying patches. my current linux comprehension ends at cloning the current gvsip git and compiling for the most part. |
| 09:15.38 | wtf911 | i appreciate you sharing the contact info. it can't hurt to ask i suppose. |
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| 13:31.14 | le_gid | Hey everybody. I was wondering if there is a way to use SIP INFO from asterisk dialplan? |
| 13:35.18 | file | there isn't |
| 13:36.29 | le_gid | so is there any other way to send INFO messages to an other server? |
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| 13:37.09 | file | chan_sip may have a dialplan application for it, but you wouldn't be able to receive it |
| 13:37.49 | file | it would be on the wiki documentation if it exists |
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| 13:48.09 | vipul20 | Hi, I am trying to implement an automated call flow using ARI. |
| 13:48.09 | vipul20 | The documentation is quite good and I was able to implement it and used |
| 13:48.09 | vipul20 | 2 softphones to test it out. But, I am facing issue on recording voice over |
| 13:48.09 | vipul20 | a call using a GSM Gateway: |
| 13:48.09 | vipul20 | app.c:1636 __ast_play_and_record: No audio available on SIP/gsmtest-00000022?? |
| 13:48.09 | vipul20 | Solutions I explored: verified NAT configurations (using force_rport,comedia), |
| 13:48.09 | vipul20 | RTP conf (using: rtpstart=10000, rtpend=20000), verified localnet and externalip. |
| 13:48.10 | vipul20 | With current configuration, I am able to record, playback etc. using softphones. |
| 13:48.10 | vipul20 | Is there anything else I should take into consideration when I try to interface with |
| 13:48.11 | vipul20 | GSM gateway? What can be a possible issue? Any leads would help a lot. Thanks :) |
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| 13:53.39 | [TK]D-Fender | Go look at the call |
| 13:53.59 | [TK]D-Fender | "GSM gateway" isn't a reason for a lack of audio. |
| 13:54.07 | [TK]D-Fender | How you configure it and what's negotiated is. |
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| 14:59.59 | vipul20 | @[TK]D-Fender: Thanks for the help. "Go look at the call" should I start by looking at SIP debug logs? What else I can look for? I know its difficult to pin point but any suspicions? |
| 15:00.17 | [TK]D-Fender | No, it isn't difficult |
| 15:00.21 | [TK]D-Fender | No audio means no audio. |
| 15:00.32 | [TK]D-Fender | and that means looking at the SIP negotiation |
| 15:05.12 | vipul20 | ok, got it. Sure I'll check that. |
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| 15:39.02 | igcewieling | Is thee anyway in the GUI to prevent specific voicemail passwords from being set in the GUI? I already know how to do that for users changing their voicemail password using VoiceMailMain |
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| 15:43.19 | Samot | There's no GUI in Asterisk. |
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| 15:50.00 | Samot | In regards to FreePBX, no. |
| 15:52.22 | Samot | The only thing the GUI does it make sure the input is 0-9 but it doesn't look for any other patterns or have any sort of checks in place. |
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| 16:31.15 | igcewieling | Samot: Thanks. I guess I'l have handle it in the traditional way: complain to management about the tech doing the stupid stuff. |
| 16:31.32 | igcewieling | I'd hoped to have the system simply not allow 1235 as a password. 8-| |
| 16:32.27 | Samot | Do most PBXes stop that? |
| 16:35.08 | Samot | Do you end users know they are on Asterisk/FreePBX systems? |
| 16:35.29 | Samot | Or do they just care that they have a "hosted voice/pbx" solution for their phones to connect to? |
| 16:40.10 | Samot | Sorry but I just find it a bit amusing when ITSPs/Voice Providers install a Business Level PBX as their solution and then complain or do the "I hoped/wished for X" about it because it doesn't meet or consider provider level policies/use/rules. |
| 16:40.30 | Samot | It's designed as an on-premises company PBX. |
| 16:41.33 | Samot | Based on the amount of commenting you've done on people trying to use simple VM passwords you should know that it's not a high priority/security issue for the average company. |
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| 17:44.24 | igcewieling | Samot: I don't know what other PBXs do. I'm trying to solve the problem on the PBXs I manage. |
| 17:45.54 | igcewieling | You are correct, it isn't a high priority for most companies. That does not mean our dickhead tech should set bad passwords. |
| 17:46.08 | Samot | Sure. |
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| 18:52.22 | Kobaz | Samot: half our customers use 1234 as a voicemail password |
| 18:54.15 | drmessano | That really wasn't his point |
| 18:54.30 | Kobaz | i was just throwing that in therer |
| 18:55.53 | drmessano | The OP wanted to force policies on VM users, and Samot's point was "Don't deploy FreePBX and some high end security policies for VM users" |
| 18:56.02 | drmessano | The OP wanted to force policies on VM users, and Samot's point was "Don't deploy FreePBX and expect some high end security policies for VM users" |
| 18:57.06 | Samot | Kind of. |
| 18:57.40 | Samot | It was more "Don't deploy an on-premises designed business level PBX as a provider solution" |
| 18:57.59 | drmessano | well, that's the same thing |
| 18:57.59 | Samot | "And expect provider level policies in it" |
| 18:58.04 | Samot | Yeah. |
| 18:58.25 | drmessano | FreePBX isn't an ITSP-In-A-Box |
| 18:58.42 | drmessano | It's literally Free->PBX |
| 18:59.19 | drmessano | But then again |
| 18:59.39 | drmessano | FreePBX + Google Voice + Whole buncha patches = Money Printing Machine |
| 19:01.00 | drmessano | If you deployed FreePBX with GV more than 6 months ago and aren't driving a Lambo yet, you suck at life |
| 19:04.48 | robmal | Damn. |
| 19:04.52 | drmessano | Samot: https://usercontent.irccloud-cdn.com/file/WddCgJrj/IMG_0851.JPG |
| 19:05.07 | Samot | HAHAHAHAHAHAHAHA |
| 19:06.10 | Samot | Aaannd that's a "right click --> Save Image As" |
| 19:06.38 | drmessano | Glad you approve |
| 19:06.48 | Samot | Highly. |
| 19:08.28 | drmessano | We were talking this week about the commercial for the TV antenna that you stick on the wall, where they have a guy blacked-out that's supposed to be a broadcast engineer, and he's talking about how the TV networks are required by law to supply a hidden over the air signal that you can receive for free, and it's a secret the cable companies dont want you to know about |
| 19:08.48 | drmessano | Thats what OTA TV is now to millenials.. a secret signal you can get with a bootleg antenna... |
| 19:09.15 | drmessano | So yeah.. GV is clandestine telephony.. and telco's are killing people that talk about it |
| 19:10.08 | Samot | Uhm. |
| 19:10.35 | Samot | Just about any tv with an antenna can pickup the signals being broadcast over the airwaves. |
| 19:10.44 | drmessano | SHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHH |
| 19:10.57 | drmessano | CABLE COMPANIES DONT WANT ANYONE TO KNOW THIS |
| 19:11.25 | Samot | Well I doubt TruTv is riding airwaves. |
| 19:13.12 | Samot | "This antenna will let you listen to music being broadcast over the airwaves. Spotify, Amazon, Pandora and other music streaming services want to keep this secret from you" |
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| 19:15.05 | drmessano | Exactly |
| 19:15.43 | Samot | I'll consider Google Voice a viable threat when they have the little things... |
| 19:15.46 | Samot | Like support. |
| 19:16.01 | drmessano | An actual pubic endpoint? |
| 19:16.04 | drmessano | public |
| 19:16.07 | drmessano | or pubic too |
| 19:16.09 | Samot | Sure. |
| 19:16.17 | Samot | Or, 911 services. |
| 19:16.34 | Samot | Hell, even a guarantee/SLA for service. |
| 19:16.44 | drmessano | Sure, but what about an actual published, public interface like most legitimate services? |
| 19:17.00 | drmessano | Oh I forgot something else |
| 19:17.01 | Samot | That's a start. |
| 19:17.06 | drmessano | What about an actual published, public interface like most legitimate services? |
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| 19:17.34 | drmessano | The hacky thing was fun, and great work by those who tackled it |
| 19:17.48 | drmessano | But, it was a throwaway service |
| 19:18.57 | adeel | can i add a sip trunk as a call queue endpoint? that is, the agents live on a far end of sip trunk behind 1 username? |
| 19:19.12 | drmessano | I really wanted to start a tip jar.. for every person that came in here "MY GV IS DOWN, HELP.. I CANT MAKE ANY CALLS AND I AM ___<insert some condition that requires urgent access to telephony services>__" |
| 19:19.42 | Samot | adeel: You could be it would be pointless. |
| 19:19.50 | adeel | why's that? |
| 19:20.00 | Samot | adeel: The queue would only see that sip trunk as the "agent" |
| 19:20.16 | adeel | Samot: it would only see it as 1 agent? or as many calls as the agent can take? |
| 19:20.27 | Samot | 1 agent. |
| 19:20.35 | adeel | i guess a better way to ask this question, can i send multiple calls to the "same" agent? |
| 19:20.40 | adeel | via queues |
| 19:20.51 | Samot | Of course |
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| 19:22.39 | Samot | But if this is for that "many agents behind one sip trunk" option... |
| 19:22.52 | Samot | The queue will never know when there are actual agents available. |
| 19:23.19 | adeel | well, there's a full ACD system at the far end of the sip trunk |
| 19:23.35 | adeel | i just need to control the flow to that ACD to prevent it from getting overloaded (it's a legacy system) |
| 19:23.50 | Samot | So why do you need a queue for this? |
| 19:23.58 | Samot | Just send calls down the sip trunk |
| 19:24.02 | adeel | to hold the calls active |
| 19:24.20 | Samot | It won't hold active calls. |
| 19:24.36 | Samot | Not like you think. |
| 19:24.50 | adeel | when the sip trunk is full, e.g. the far end returns a 486 busy, i need the call to not terminate....i could probably do something in dialplan after the DIAL() to catch those calls |
| 19:24.56 | Samot | If you have a single agent (aka the sip trunk) and you have it sends calls regardless of being on a call or not.. |
| 19:24.58 | adeel | but then i need it to retry shortly after |
| 19:25.09 | Samot | It will send the calls as soon as they enter the queue on the next agent ring cycle. |
| 19:28.05 | Samot | Does this other PBX have some sort of limit to how many channels it can support on a sip trunk? |
| 19:28.22 | adeel | yeah it does |
| 19:28.47 | Samot | OK, so the queue is not going to control that. |
| 19:29.09 | Samot | It's going to send them to the PBX, it's going to return a 486 and it will bounce them back to the queue. |
| 19:29.14 | adeel | well, the queue will handle the retry attempts....if the far end responds with a 486 busy, then the queue will reschedule the attempt |
| 19:29.45 | Samot | So the plan is to hold them in a temp queue |
| 19:29.54 | Samot | Until the main queue can take them and hold them... |
| 19:29.59 | adeel | yup, pretty much |
| 19:30.07 | Samot | Alright |
| 19:30.17 | adeel | i know i can write a dialplan to do this, but the queue module seemed more elegant |
| 19:31.21 | Samot | Well you should make sure the MoH matches on each system... |
| 19:31.36 | adeel | no MOH, just ringing |
| 19:32.02 | Samot | How long do you expect them to sit there listening to ringing? |
| 19:32.10 | Samot | People do have a limitation for that... |
| 19:32.22 | Samot | Endless ringing makes it seem like there is nothing there to answer. |
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| 19:34.07 | adeel | Samot: that's a customer/end user problem. not my concern. my job is to build the flow |
| 19:34.13 | adeel | i appreciate it, but i don't really care |
| 19:34.48 | Samot | Oh so this is a project job. |
| 19:34.59 | Samot | Not your company PBX. |
| 19:36.31 | adeel | those details dont' matter; my original question of can a queue send multiple calls to a single endpoint is the only relevant part |
| 19:38.18 | adeel | n/m, just found the answer...set ringinuse=yes and asterisk will send multiple calls to the endpoint |
| 19:38.51 | [TK]D-Fender | <PROTECTED> |
| 19:39.38 | adeel | so what happens when the far end point sends a 486, with ringinuse=yes ? does * re-queue the call and re-attempt it a few seconds later? |
| 19:39.55 | adeel | assuming a single agent in the queue |
| 19:40.26 | Samot | I explained what would happen |
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| 19:41.20 | adeel | "Samot: It's going to send them to the PBX, it's going to return a 486 and it will bounce them back to the queue." ....and what happens after they're bounced back to the queue? |
| 19:41.47 | adeel | is the position lost? does it retry immediately, etc |
| 19:41.56 | seanbright | "< Samot> Until the main queue can take them and hold them..." |
| 19:42.17 | seanbright | just test it |
| 19:42.19 | seanbright | what's the issue here? |
| 19:42.23 | [TK]D-Fender | <adeel> is the position lost? does it retry immediately, etc <- it follows queue distribution just like normal |
| 19:42.58 | Samot | 3:28:46 PM <Samot> OK, so the queue is not going to control that. |
| 19:42.58 | Samot | 3:29:08 PM <Samot> It's going to send them to the PBX, it's going to return a 486 and it will bounce them back to the queue. |
| 19:43.11 | seanbright | the answer to any question about 'app_queue' is 'probably, but it is going to be ugly' |
| 19:43.25 | [TK]D-Fender | Call enters queue. Selects agents. Rings them for allotted time (max). On failure or timeout of ringing it will WAIT the amout of time you told it to. |
| 19:43.48 | Samot | They do not lose their position. |
| 19:43.58 | Samot | They would still be caller 1 if the agent didn't answer. |
| 19:44.05 | [TK]D-Fender | you never bounce back to the queue... you never left if it only 486's |
| 19:44.18 | [TK]D-Fender | or anything that isn't counted as an "answer" |
| 19:44.22 | Samot | ^^ technically yes.. |
| 19:44.27 | adeel | k, thanks |
| 19:44.43 | [TK]D-Fender | which.... includes VOICEMAIL on a remote end, etc... better nothave INBAND responses there... or have enabled some sort of confirmation setup... |
| 19:44.58 | [TK]D-Fender | Things to account for... |
| 19:47.20 | *** part/#asterisk seanbright (~sean@asterisk/community-developer/seanbright) |
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| 19:47.23 | *** mode/#asterisk [+o seanbright] by ChanServ |
| 19:47.54 | *** topic/#asterisk by rmudgett -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12), Standard: 15.5.0 (2018/07/12); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 19:49.31 | Samot | [TK]D-Fender: We're not accounting for those things. This is a "temp" queue for with the PBX with the real queue has to many calls for the SIP trunk to accept new calls. |
| 19:49.40 | Samot | The users are just going to sit there hearing ringing. |
| 19:49.52 | Samot | Hopefully those aren't going to be long calls. |
| 19:50.24 | [TK]D-Fender | Wel he said he was calling out. |
| 19:50.26 | *** join/#asterisk hans109h (~jw__@h184-60-28-43.mdtnwi.dsl.dynamic.tds.net) |
| 19:50.30 | Samot | Yeah. |
| 19:50.35 | [TK]D-Fender | Is the "out" technically to "nowhere" and no real end? |
| 19:50.37 | Samot | From Asterisk to the PBX with the real ACD |
| 19:50.44 | Samot | The agents are on that PBX |
| 19:50.47 | [TK]D-Fender | so a queue to a queue? |
| 19:50.49 | Samot | Yes. |
| 19:50.53 | [TK]D-Fender | WTF |
| 19:50.55 | Samot | In a sense. |
| 19:51.01 | Samot | Because the SIP trunk has X channels. |
| 19:51.09 | Samot | So new incoming calls are getting busy |
| 19:51.19 | [TK]D-Fender | Is this to use a queue to limit the channels? |
| 19:51.34 | drmessano | uh oh |
| 19:51.37 | Samot | Or hold them until capacity is lowered. |
| 19:51.44 | drmessano | 13.22 is out. GVSIP just broke again |
| 19:51.48 | [TK]D-Fender | This sounds pretty sad.... |
| 19:52.08 | [TK]D-Fender | drmessano, You could leave that first sentence out entirely on a continuing basis... |
| 19:52.16 | hans109h | I've tried 2 different ways to use the SayAlpha command to spell out the content of a variable, but the way that it is working is that it just spells the name of the variable. Here is one example: same = n,SayAlpha(${PublicIP}) |
| 19:52.19 | adeel | [TK]D-Fender: it does a few other functions before the queue portion, but yeah |
| 19:52.24 | hans109h | is it a syntax problem? |
| 19:53.18 | [TK]D-Fender | hans109h, Show us the call.... |
| 19:53.34 | [TK]D-Fender | hans109h, CLI tells all.... |
| 20:25.36 | hans109h | here is the CLI portion of the other way I tied to do it: https://pastebin.com/2b0ntMsz |
| 20:28.49 | [TK]D-Fender | that of course doesn't look like the previous bit you showed us |
| 20:28.51 | [TK]D-Fender | "$(SHELL(curl -kLs http://api.ipify.org))") |
| 20:29.05 | [TK]D-Fender | $(SHELL <- and the failure is pretty clear. wrong brace |
| 20:29.38 | [TK]D-Fender | NEXT!@@@!@!! |
| 20:50.50 | *** join/#asterisk jamesaxl (~James_Axl@41.140.230.87) |
| 20:57.23 | [TK]D-Fender | BBIAB |
| 20:58.40 | hans109h | That's why I said it was the other way I tried to do it, either way it isn't working for me, but from your exclamation it appears there is a problem with my shell function so I will look at that. |
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| 21:06.54 | drmessano | hans109h: Any reason you're not using app_curl for this? |
| 21:08.02 | *** join/#asterisk cr1 (~cr@ip174-68-116-216.sd.sd.cox.net) |
| 21:08.38 | hans109h | drmessano, come on, I just got it working this way and now you said that! :-) just kidding....only because it's been a few years since I've made any changes to Asterisk so I'm pretty rusty. |
| 21:09.13 | drmessano | I think CURL() has been around for 10 years or so |
| 21:12.20 | hans109h | So the benefit of using app_curl vs a shell is lower resource consumption? |
| 21:13.36 | *** join/#asterisk bouncer151 (62c27807@gateway/web/freenode/ip.98.194.120.7) |
| 21:14.44 | drmessano | Cleaner way of doing it |
| 21:17.55 | drmessano | exten => 23,1,SayAlpha(${CURL(http://api.ipify.org)}) |
| 21:17.59 | drmessano | hans109h: ^ |
| 21:20.42 | hans109h | Thanks for the suggestion, I was just finishing up doing exactly that. Works quite nice. |
| 21:21.21 | bouncer151 | hi, looking to see if I can get some clarification on getting asterisk to output through soundcard for an overhead page. I have attempted to load chan_alsa.so but it fails to load and the reason is because the soundcard cannot match to 8000hz for sample rate. Has anybody successfully gotten paging to work out of asterisk? I read somewhere it "used" to work and now no longer works. |
| 21:22.07 | hans109h | I decided to set the curl response as a variable, so that I can repeat the IP address twice without having to make two calls to ipify.org, seemed cleaner. |
| 21:35.06 | hans109h | Thanks for the help. Good night. |
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