| 00:18.47 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:18.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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 |
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| 00:34.09 | jaredkipe | Hello all! |
| 00:35.26 | jaredkipe | I could use some help. I've been working on an asterisk 14.4.0 docker for a little while now. Everything has been going really well until I went to try to do some TLS web sockets. It doesn't give any sort of error or explanation, but it certainly does not bind to the port I've told it to. |
| 00:36.11 | jaredkipe | Is there anything build specific to make this work? (I've also tried 13.15 with the same effect, no log, but no HTTPS binding either.) |
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| 01:05.54 | CRCinAU | hrrrm ok |
| 01:05.58 | CRCinAU | so I do have one more problem. |
| 01:06.16 | CRCinAU | When trying to Dial(PJSIP/901), I end up with: |
| 01:06.29 | CRCinAU | digest_create_request_with_auth: Unable to create request with auth. No auth credentials for any realms in challenge. |
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| 01:07.43 | CRCinAU | and 'realm' is blank in pjsip show auth 901 |
| 01:12.24 | CRCinAU | so I'm guessing this is another difference between chan_sip and pjsip |
| 01:14.41 | jaredkipe | Has anyone ever gotten http +tls to work? |
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| 02:51.25 | Samot | karedkipe: Yes. |
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| 03:40.30 | igcewieling | sip scanners are out in force tonight. |
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| 05:02.01 | Maliuta_ | it's not just sip. My smtp is being hit hard too |
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| 05:47.32 | drmessano | It's cyber warfare |
| 05:47.44 | drmessano | Hightened tensions right now |
| 05:48.07 | drmessano | Usually see an uptick after someone pisses in someone elses Cheerios |
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| 06:35.03 | Maliuta_ | Considering who is living in the white house ... |
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| 07:25.36 | drmessano | or Pyongyang or Damascus |
| 07:26.15 | drmessano | Sorta silly to blame all the worlds problems on one guy |
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| 12:00.27 | user981432 | Hey, I'm currently doing my first steps on Asterisk here as a small PBX. I'm currently wondering why Asterisk uses a random source port when sending REGISTER messages using TLS/TCP - normal? |
| 12:02.20 | user981432 | I mean, my soft- or deskphones use the SIP listening port as a source port for SIP messages |
| 12:07.51 | file | if using UDP it would be the same, if using TCP/TLS it would be a random source port as that is how TCP and TLS works |
| 12:08.15 | file | the operating system gives the connection a local ephemeral port |
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| 12:09.34 | user981432 | file: Hm, okay, makes sense because of TCP session tuples |
| 12:10.40 | user981432 | file: If using TCP, how do you make sure Asterisk is reachable for SIP traffic from the upstream registrar? Do I explicitly need to open port 5061 locally? |
| 12:11.21 | file | that depends on the behavior of the remote upstream... the Contact specifies the IP address+port to contact back on, so you could need that open |
| 12:11.27 | file | but they may also reuse the existing connection if it is up |
| 12:13.06 | user981432 | Hm, Asterisk sends the local port 5061 as a contact |
| 12:13.26 | file | yes, because it expects a connection back on that port |
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| 12:14.26 | user981432 | So I'd need to make sure the TCP connections stays up if I have any hopes of NOT needing to open a local port. |
| 12:14.41 | file | right. |
| 12:14.49 | file | and that assumes the remote side will reuse the connection. |
| 12:15.16 | user981432 | Can I strip the contact header? |
| 12:15.32 | user981432 | I mean, it should be RFC-compliant, shouldn't it? |
| 12:16.25 | file | strip the contact header? Contact is mandatory |
| 12:16.37 | user981432 | file: Are you sure? On a REGISTER? |
| 12:16.41 | file | yes. |
| 12:16.46 | user981432 | https://tools.ietf.org/html/rfc3261#page-57 |
| 12:17.28 | user981432 | Is this RFC superseded? |
| 12:17.36 | file | no |
| 12:17.44 | user981432 | Then it isn't mandatory |
| 12:18.19 | user981432 | Okay, that sounds bitchy. I really meant to ask whether it is the current RFC |
| 12:18.20 | file | if you want to actually add a Contact to be reachable... it is |
| 12:19.05 | file | lack of a Contact header would return a 200 OK with current bindings, it would not actually do a registration |
| 12:20.45 | user981432 | Hm, yes |
| 12:21.03 | user981432 | That IS annoying. |
| 12:21.12 | file | that is the way SIP registration works. |
| 12:21.27 | user981432 | Yeah, I've only so far used UDP |
| 12:23.37 | user981432 | DTLS ftw :( |
| 12:25.28 | user981432 | Thank you for your help! |
| 12:25.38 | file | yw |
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| 14:13.36 | dnit | Hi |
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| 14:15.44 | dnit | Hi anyone there ? |
| 14:16.03 | Samot | No. |
| 14:16.15 | dnit | LOL |
| 14:16.27 | dnit | Can I get a quick help. |
| 14:18.23 | Samot | No. |
| 14:18.44 | Samot | Unless you're going to ask the question about the help you need... |
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| 14:23.34 | Samot | ? |
| 14:24.24 | dnit | Sorry I was exploring a bit more before asking. |
| 14:24.57 | dnit | I am getting error: pbx.c:4489 ast_func_write: Function PJSIP_HEADER not registered |
| 14:25.08 | Samot | Are you using PJSIP? |
| 14:26.24 | dnit | I am using PJSIP_HEADER(add,some)=${someothervar}) |
| 14:26.33 | dnit | I dont't think I am using PJSIP |
| 14:26.45 | Samot | Then why are you using a PJSIP function? |
| 14:26.50 | dnit | My astyerisk version is 13.6 |
| 14:26.59 | Samot | That doesn't clarify. |
| 14:27.07 | Samot | There is Chan_SIP and Chan_PJSIP. |
| 14:27.50 | dnit | Yes, but chan_PJSIP has completely different syntax for sip.conf right ? |
| 14:27.52 | Samot | It say it's not registered probably means that The chan_pjsip module isn't loaded. |
| 14:27.57 | Samot | No. |
| 14:28.06 | Samot | Because PJSIP doesn't go in sip.conf |
| 14:28.14 | Samot | It goes in pjsip.conf |
| 14:28.16 | dnit | How can I load it ? |
| 14:28.32 | Samot | The same way you load any other modules. |
| 14:29.27 | Samot | module load chan_pjsip.so |
| 14:30.09 | Samot | And then make sure modules.conf has the right settings to load chan_pjsip.so |
| 14:30.39 | Samot | But if you make chan_sip peers/trunks/extensions then using PJSIP functions are pointless. |
| 14:32.15 | dnit | Yes I think I need to do more research, since our production servers are workin on the same code I pulled. :( |
| 14:36.30 | Samot | Well is your production server a pure asterisk server? |
| 14:36.39 | Samot | Is it running PJSIP? |
| 14:36.55 | dnit | yes asterisk 13.6 |
| 14:37.02 | Samot | Just asterisk? |
| 14:37.12 | Samot | It's not FreePBX or some other distro using Asterisk? |
| 14:37.27 | dnit | Sorry not sure what you mean by just asterisk. Its not freepbx |
| 14:37.56 | dnit | Yes just asterisk. |
| 14:38.00 | Samot | I mean, you're not using some distro that has a GUI or other features that use Asterisk. |
| 14:38.17 | Samot | Well you should compare what is actually running and installed on the production server and make sure the new server is doing the same. |
| 14:38.28 | Samot | Just pulling code doesn't mean it's going to just work. |
| 14:40.16 | dnit | You are right, I am just afraid of touching prod server. |
| 14:40.35 | Samot | You just to look. |
| 14:40.58 | Samot | If you don't know if it's running PJSIP or not and you write things for PJSIP that's kind of a waste, don't you think? |
| 14:42.01 | dnit | Asterisk certified/13.8-cert2 this is what I have on prod. |
| 14:46.09 | Samot | That doesn't really mean anything. |
| 14:46.16 | Samot | It's the version of Asterisk you are using. |
| 14:46.36 | Samot | That doesn't tell me anything about what channel drivers are in use, what other features or functions are being used. |
| 14:46.49 | dnit | I am sorry how can I verify its using pjsip and not chansip ? |
| 14:46.52 | Samot | If you're building a new system to replace the production system, you need to know what the production system is dong. |
| 14:46.58 | Samot | Look at the configs. |
| 14:47.16 | Samot | It's your server, you're going to have to look at how it was configured. |
| 14:48.45 | wdoekes | asterisk -nrx 'module show' | grep sip |
| 14:49.33 | wdoekes | but you should know soon enough after looking in sip.conf and pjsip.conf; you're unlikely to have production config in both |
| 14:50.40 | dnit | Yes unfortunately there is configuration in both the files, and all this time I though we werent using pjsip |
| 14:50.49 | dnit | Prod is using pjsip. |
| 14:51.33 | dnit | Do I have to build asterisk from source again to include pjsip ? |
| 14:52.15 | dnit | Becuase I can't do module load chan_pjsip.so |
| 14:52.33 | dnit | It throws error. |
| 14:55.56 | Samot | You need to compile Asterisk with PJSIP |
| 14:56.21 | dnit | Thanks for your help. |
| 14:56.30 | dnit | Appreciate it! |
| 14:56.37 | dnit | Have a good day! |
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| 16:50.30 | canthus13 | I've got a customer who wants to be able to essentially pick through the parking lot from any phone and return the calls back to the lot with one button push. The closest I can come up with is to throw several extensions into a ring group, add them to line keys, and then he can switch around through the line keys at will, returning them to hold when he selects the next line. Unfortunately, I can't find a way to allow another phone to also |
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| 16:59.10 | igcewieling | canthus13: that is all possible, except for the one button push |
| 16:59.53 | igcewieling | You could add a line key for each parking lot slot, but not for each call currently in the parking lot. |
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| 17:27.14 | igcewieling | heh, I just realized none of the Asterisk sounds files include the word "toll" or the words "long distance" |
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| 17:43.51 | malcolmd | :D what is this interlata dialing of which you speak? |
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| 18:01.39 | canthus13 | igcewieling: I've got line keys for each slot... but that's too many keypresses for him. And apparently nobody at Sangoma thinks it's doable, either. So.. this dude will probably quit us, go to comcast, and find out that they're gonna run into the same thing: He's too picky. |
| 18:02.20 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 18:02.52 | igcewieling | canthus13: Your only option at this point is to swap out the damaged user with an undamaged one. |
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| 19:04.31 | canthus13 | igcewieling: Pretty sure he's gonna go be comcast's problem soon, so... |
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| 21:13.36 | *** join/#asterisk hdon (~hdon@68.110.137.138) |
| 21:14.10 | hdon | hi all :) this is going to sound a little silly: is there such a thing as a phone call? |
| 21:16.02 | Samot | No, not at all. |
| 21:16.16 | hdon | i didn't think so |
| 21:16.31 | hdon | it seems like something phone companies invented for billing purposes |
| 21:16.37 | Samot | I mean before Bell figured it out. |
| 21:16.40 | hdon | but in reality there are only bridges and channels |
| 21:16.57 | hdon | :C |
| 21:17.06 | Samot | What bridges and channels? |
| 21:17.38 | hdon | these |
| 21:17.41 | hdon | gestures vaguely |
| 21:17.45 | hdon | see? |
| 21:29.36 | drmessano | Are you ON something? |
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| 21:32.43 | hdon | forgive me drmessano, i'm an asterisk newb |
| 21:32.44 | hdon | suppose Alice calls Bob. then Cindy joins. then Alice leaves. then Dennis joins. then Bob leaves. is this one "phone call?" |
| 21:33.45 | hdon | is it possible to have a more exotic type of bridge that can split into two bridges that somehow divides up the channels of the original bridge between them? |
| 21:34.08 | hdon | such a thing is at least conceivable to me, and it is not obvious to me where the "phone call" or "phone calls" exist in such a scenario |
| 21:34.15 | Samot | OK.. |
| 21:34.23 | hdon | does this make a little more sense? |
| 21:34.28 | Samot | Party A calls DID |
| 21:34.42 | Samot | DID is routed via the PSTN to your endpoint, which in this case is Asterisk. |
| 21:35.13 | Samot | Asterisk sends that call directly to your phone... |
| 21:35.24 | Samot | Party A is now in a call with Party B |
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| 21:35.40 | Samot | That is a call. |
| 21:35.58 | hdon | ok but you have described what i think is the simplest use case of a phone system |
| 21:36.07 | hdon | what about in more complex scenarios? |
| 21:36.15 | Samot | A phone system is not needed for that scenario. |
| 21:36.17 | Samot | At all |
| 21:36.39 | Samot | Party B wants to make and receive calls on the PSTN. |
| 21:36.44 | Samot | No PBX is needed for that. |
| 21:36.51 | klow | Anyone know why "pjsip show channelstats" would say "ulaw" when I call the echo test, even though the softphone says OPUS, and ulaw isnt even in the list of allowed codecs anywhere ? |
| 21:37.00 | hdon | mmm |
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| 21:37.09 | klow | Using pjsip and OPUS and getting two-way audio on calling echo test, but nothing between 2 phones. |
| 21:38.09 | Samot | klow: I don't believe there is a transcoding path for opus <--> ulaw |
| 21:39.22 | hdon | hmm |
| 21:39.38 | hdon | ok, well, setting aside the more existential questions i have for the moment |
| 21:39.40 | Samot | hdon; What are you trying to do? |
| 21:39.47 | hdon | i do have a more direct question |
| 21:39.55 | hdon | how is call recording facilitated in asterisk? |
| 21:40.15 | Samot | Asterisk has a few methods to record calls. |
| 21:40.28 | Samot | You would have to determine what is the best option for what you need. |
| 21:40.44 | hdon | is there a detailed overview of all these methods some place?\ |
| 21:41.15 | Samot | Yes, the Asterisk wiki and the sample config files for Asterisk. |
| 21:43.10 | *** join/#asterisk Samael28 (~Samael28@79.110.128.128) |
| 21:43.14 | k3asd` | hi there, how can I understand why my pbx does not receive any calls? |
| 21:44.20 | k3asd` | I set up the sip and extensions section but when I use my phone to call a number configured the calls dropped |
| 21:44.28 | k3asd` | if you want I can share my conf |
| 21:50.22 | hdon | lots of instances of "record" in the configuration samples are not about recording :3 |
| 21:51.27 | hdon | or, not the kind of recording i want |
| 21:51.42 | hdon | lots of CDR-related matches |
| 21:51.49 | hdon | a few DNS-related matches |
| 22:01.38 | Samot | You are asking vague questions. |
| 22:02.24 | igcewieling | The answers you seek are already within your reach, grasshopper. |
| 22:05.20 | hdon | Samot, yes i know |
| 22:05.37 | hdon | sorry |
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| 23:14.46 | *** join/#asterisk Geriatrix (~kvirc@207.194.15.162) |
| 23:15.39 | Geriatrix | hey guys - how do i install asterisk on centos via package manager these days - i used to use asterisk now and maintain it with yum but seems that's not an option any more - i need a new install but i dont' want freepbx |
| 23:31.31 | hdon | Geriatrix, http://packages.asterisk.org/centos/ ? |
| 23:31.40 | hdon | Geriatrix, looks out of date though |
| 23:31.59 | file | we do not currently maintain packages |
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| 23:35.34 | hdon | Samot, ok so, my issue is that, whenever i transfer a call, the recording ends |
| 23:35.52 | hdon | i'm given to understand this is kind of a sore spot for asterisk admins |
| 23:35.55 | hdon | is that true? |
| 23:43.38 | Geriatrix | hdon: yes - seems so - there has to be a way to install a package and have it updates and maintained without having to compile each time ? |
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| 23:51.03 | hdon | Geriatrix, someone needs to maintain those updates for you. i couldn't tell you who might offer those for your platform. sorry |
| 23:52.13 | Geriatrix | but digium used to do that - is that no longer the case ? |
| 23:56.23 | file | Digium does not maintain packages currently |
| 23:58.04 | Geriatrix | :( thats a bummer :) |
| 23:58.13 | Geriatrix | what's the best practise then ? |
| 23:58.28 | Geriatrix | i dont' need freepbx |
| 23:58.38 | Geriatrix | but woudl like packaged version of asterisk :) |