| 00:29.33 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:1007:eb79:a095:527:42c5:5fde) |
| 01:01.31 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 01:19.40 | igcewieling | looks like a stuck call |
| 01:26.06 | *** join/#asterisk giesen (~ggiesen@ego.giesen.me) |
| 01:28.24 | Nivex | no, it's actually been running that whole time |
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| 03:03.02 | igcewieling | what version of Asterisk? |
| 03:06.14 | igcewieling | older versions of Asterisk don't track that sort of stuff very well. |
| 04:38.57 | Zombie | Anyone still here? |
| 04:40.39 | drmessano | There's 165 nicks in here |
| 04:40.51 | drmessano | We don't do roll calls |
| 04:40.57 | drmessano | If you have a question, ask |
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| 04:55.27 | Zombie | Well my lunch is almost up. |
| 04:56.04 | Zombie | I was going to ask about doing station to station calls between my Android Phones and my Cisco SPA303 |
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| 04:58.41 | drmessano | What about it? |
| 05:00.02 | drmessano | Dude, if you have an issue or an actual question, ask. This is IRC, there's no number to take and wait on like the Deli. JFC |
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| 05:53.15 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.32.0 (2020/03/12) 16.9.0 (2020/03/12) Standard: 17.3.0 (2020/03/12); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 11:27.59 | Zombie | Anyone still awake? |
| 11:29.18 | Nivex | nah, I just woke up: https://xkcd.com/448/ |
| 11:36.02 | *** join/#asterisk AsteriskRoss (~AsteriskR@r01.nt-r1.nor.gb.voicehost.co.uk) |
| 11:36.35 | Zombie | I'm trying to deal with a project dealing with Asterisk., |
| 11:37.18 | Zombie | At this Phase, I just want my SIP Phones to be able to call each other. |
| 11:37.47 | Zombie | I eventually want to get a SIP Provider for the outside world, |
| 11:37.58 | Zombie | but one step at a time. |
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| 12:37.11 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 12:42.02 | Samot | What kind of project with Asterisk? |
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| 14:25.36 | Zombie | Samot, Home use. |
| 14:26.35 | Samot | OK. What is the goal of the project? |
| 14:31.35 | *** join/#asterisk allizom (~Thunderbi@unaffiliated/allizom) |
| 14:34.01 | Zombie | A few goals: |
| 14:35.00 | Zombie | Android Phones, and my Cisco Base Station should be able to call each other. |
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| 14:35.17 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:35.23 | allizom | Hi, when setting up local extensions, what should the actual numbers look like not to interfere with public numbering plans? I've checked in my country numbering plan any extension starting with either 2, 6 or 9 is "reserved for future use", is there a prefix set apart in most countries for this kind of use? |
| 14:35.52 | Zombie | 2. I Have an existing XMPP setup. I'd like to be able to use the XMPP, and LDAP Setup too: |
| 14:37.19 | Zombie | 2a. Recieve a call intended for my Sim-Carded Android Phone on either Ekiga Soft Phone or the Cisco SPA303 Base Station. |
| 14:38.43 | Zombie | 2b. Have the Phone upload Voice Mail to Asterisk Servers. |
| 14:43.48 | Samot | In what way do you want to use this XMPP setup with Asterisk? |
| 14:44.59 | *** join/#asterisk cresl1n (uid299068@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:45.00 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:45.46 | [TK]D-Fender | allizom, they should not looks like public numbering plans... |
| 14:45.53 | [TK]D-Fender | allizom, Your question answers itself |
| 14:46.40 | allizom | [TK]D-Fender: what I'm asking is whether there is a specific prefix set apart and explicitly intended to be used in this way |
| 14:46.56 | allizom | which most numbering plans agree on |
| 14:48.29 | igcewieling | allizom: are you in Germany? |
| 14:48.34 | allizom | nope |
| 14:48.56 | igcewieling | For the most part, extensions and phone numbers won't interfere with each other. |
| 14:49.09 | [TK]D-Fender | allizom, If you're loking for local extensions why wouldn't you just make them SHORTER than all those other patterns? |
| 14:49.21 | igcewieling | allizom: there is no standard. |
| 14:50.05 | allizom | [TK]D-Fender: I want a short extension number, I just want to make sure it's not going to conflict with other uses |
| 14:50.09 | allizom | igcewieling: I see |
| 14:50.26 | [TK]D-Fender | allizom, you should know what valid phone numbers look like. |
| 14:50.37 | [TK]D-Fender | allizom, You just invent your own prefix to dial out. |
| 14:51.05 | allizom | [TK]D-Fender: I was asking to check whether there was an agreed on prefix already |
| 14:51.12 | allizom | but there isn't. fine. |
| 14:51.29 | Samot | Sure thre is. |
| 14:51.34 | Samot | there* |
| 14:51.53 | allizom | Samot: I'd like to know more about it |
| 14:52.06 | Samot | 9, at least in the US/Canada, has been the defacto "get an outside line" prefix for decades. |
| 14:53.16 | Samot | However, prefixes that do specific things such as use X route or set Y CallerID are up to those that deploy it and therefore there is no "agreed" standard. |
| 14:55.38 | allizom | Samot: I'm not in North America and I'm not familiar with their plan, but aren't there publicly used numbers which start with 9? 911 comes to mind |
| 14:55.50 | Samot | Those are special numbers. |
| 14:56.03 | Samot | And yes there are area codes that start with 9 |
| 14:56.29 | allizom | Is there any single-digit prefix which is not used whatsoever there? |
| 14:56.43 | Samot | That is on the admin. |
| 14:57.24 | Samot | The only destination that can never be prefixed as of 2020 is 911 because that's the emergency services. |
| 14:57.45 | Samot | That _must_ be able to be dialed as is and reach emergency services. |
| 14:58.09 | Samot | However, 411 which is Information Services can conflict with a company extension plan. |
| 14:58.14 | Samot | There could be an extension 411 |
| 14:58.45 | Samot | So that means the users can't dial 411 to get to Information Services. They would have to dial something different. |
| 14:58.47 | *** join/#asterisk miralin1 (~Thunderbi@195.209.246.194) |
| 14:58.56 | Zombie | I'm not sure yet. |
| 14:58.59 | Samot | What that is, totally up to the admin of the PBX. |
| 14:59.15 | Samot | Zombie: That is something you need to sort out. |
| 14:59.36 | Zombie | I am starting in small steps. |
| 14:59.37 | igcewieling | For PBXs (in the USA) I run, I recommend extensions be in the ranges 2 - 8, no prefix for outside lines, all outside calls start with 1 or 011, except 911 |
| 14:59.57 | igcewieling | That is my personal preference. |
| 15:01.00 | allizom | I'd like not to prefix calls going outside in any way, what I'd like to do is to limit my choice of extensions I'm setting up not to conflict with public numbers |
| 15:01.24 | igcewieling | allizom: then you are making things more complicated. |
| 15:01.31 | Zombie | Zombie, Here? |
| 15:01.53 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:1007:eb79:a095:527:42c5:5fde) |
| 15:02.30 | allizom | igcewieling: it should be quite simple if there is a (preferably short) prefix no one else is using |
| 15:02.54 | igcewieling | allizom: that all depends on YOUR country dialing plan. |
| 15:03.27 | allizom | I know, and I've checked that. It says prefixes 2, 6 and 9 are "reserved for future use" as I said before |
| 15:03.34 | igcewieling | so use those. |
| 15:03.55 | igcewieling | all your phone numbers start with 1, 3, 4, 5, 7, and 8? |
| 15:03.56 | Samot | allizom: I'm not sure how 200 can conflict with someone dialing 20010456212 |
| 15:04.30 | Samot | Unless you are making the extensions the same length as the standard digit pattern you have nothing to worry about. |
| 15:04.37 | igcewieling | you don't dial a prefix like 0 to indicate a toll call like 90% of the world? |
| 15:04.52 | allizom | Samot: ok |
| 15:04.54 | Samot | The US uses 10 digit numbers. Unless I'm making 10 digit extensions, I'm never going to conflict with a "real number" |
| 15:05.14 | allizom | we have variable length numbers in some cases |
| 15:05.24 | Samot | There is a common length |
| 15:05.41 | Zombie | Samot: Can I give you some larger elements of my network infrastructure? |
| 15:05.43 | Samot | In certain parts of the US you can use 7 digits. |
| 15:05.49 | allizom | sure, I just asked if there was a surefire way |
| 15:06.00 | Samot | So again, how many 7 digit extensions would there be? |
| 15:06.15 | Samot | Are you planning on making extensions with lengths beyond 6 digits? |
| 15:06.25 | allizom | Samot: my local extensions are not going to be this long |
| 15:06.36 | Samot | Then you have nothing to concern yourself with. |
| 15:06.39 | allizom | so I should be fine, right |
| 15:06.48 | Samot | XXX isn't going to conflict with XXXXXXXX |
| 15:06.53 | Samot | Because they aren't the same. |
| 15:07.34 | allizom | igcewieling: in our country we call national numbers by their "full number" which can start with 0 |
| 15:07.37 | Samot | Zombie: You can. |
| 15:07.47 | allizom | international calls must start with 00 as i that is reserved |
| 15:07.52 | Samot | OK |
| 15:08.21 | Samot | So if it starts with 0 it has to be at least X digits in length.. |
| 15:08.38 | Samot | And then the length can vary |
| 15:09.00 | allizom | presumably, but I don't know precisely how many minimum digits |
| 15:09.07 | Samot | Again in the US (certain parts still) you can dial between 7 to 11 digits. |
| 15:09.16 | Zombie | Samot: I use Samba 4 (As a form of LDAP, and Kerberos) eGroupware, DoveCot, Fetchmail, Spectrum2, Prosody (XMPP), Project MAXS, and MythTV with Kodi. |
| 15:09.16 | allizom | there are no 4 or 5 numbers for dure |
| 15:09.21 | allizom | *sure |
| 15:09.28 | allizom | 4 or 5 digits numbers |
| 15:10.04 | Samot | Zombie: OK, so what is the plan for all that and how does Asterisk tie into it? |
| 15:10.13 | allizom | ok people, thanks all |
| 15:11.39 | Zombie | Where I live, the Telephone company largely does not offer POTS Service, they have a POTS Compatible VOIP Service that costs extra. |
| 15:12.35 | Zombie | The Telco runs Fibre to the station, and CAT5 to the houses |
| 15:13.13 | Zombie | Come next week, the very last legacy CAT3 Wiring will be replaced with CAT5. (It was non-functional to begin with.) |
| 15:14.55 | Samot | Wait, VoIP costs more than POTS? |
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| 15:34.31 | Zombie | Let me Clarify. |
| 15:35.11 | Zombie | we upgraded from ADSL at 25 Mbps to VDSL at 75 to 100 MBps. |
| 15:35.47 | Zombie | Our Analogue wiring was snipped and new CAT5 was run into the house. |
| 15:36.29 | igcewieling | You must live somewhere civilized like Europe. |
| 15:36.37 | Zombie | The Telco used a drilled out Coaxial Jack from a long gone Cable Co. Entry point. |
| 15:37.01 | Zombie | igcewieling, Canada. |
| 15:37.31 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 15:37.31 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 15:37.33 | igcewieling | Odd. Their internet industry is almost as bad as the USA |
| 15:38.52 | Zombie | We were told that We couldn't usse the DSL Filters anymore. |
| 15:39.13 | Zombie | The Telco actually damaged some drywall. |
| 15:39.25 | igcewieling | that could make sense. |
| 15:40.52 | Samot | OK, so the new service is being delivered over fiber vs copper. |
| 15:41.01 | Zombie | by trying to put some weird surface mounted sticky tape port, purtruding out the side of the Cable Jack we now use for a Terrestrial ATSC Antenna, because we wanted to keep using the |
| 15:41.14 | Zombie | Yes basically. |
| 15:41.32 | Samot | OK so now you want to use Asterisk because you don't want to pay for phone service? |
| 15:41.34 | Zombie | Its copper to our house. (RHH-45 Port.) |
| 15:41.42 | Zombie | Its copper to our house. (RJ-45 Port.) |
| 15:41.55 | Samot | I doubt they just put copper into your house for this. |
| 15:42.12 | Samot | Everyone is on a mission to remove copper as a transport. |
| 15:43.18 | igcewieling | Except Frontier, of course. 8-| |
| 15:43.40 | Zombie | They said if we wanted to use this as a POTS line, we needed to subscribe to their in house VOIP solution because these were not Analogue lines |
| 15:43.48 | Samot | OK |
| 15:43.58 | Samot | So again, you don't want to do that so you want to use Asterisk |
| 15:44.01 | Samot | Correct? |
| 15:44.19 | Zombie | Yes, we were okay with that because, we both have Cell Phones. |
| 15:44.39 | Zombie | And the land lines were a repeated Target for Fraudulent Calls. |
| 15:44.48 | Samot | So you will still need a provider for this. |
| 15:44.57 | Samot | Otherwise you just have a PBX that calls internally. |
| 15:45.00 | Zombie | think: Jim Browning level stuff. |
| 15:45.21 | Zombie | I know that, I have some questions about that. |
| 15:46.16 | Samot | OK |
| 15:46.17 | Zombie | Is it possible for a Cell Phone with an Asterisk client setup to forward a call to an Asterisk base station? |
| 15:46.39 | Zombie | lie my Cisco SPA303 or Ekiga SoftPhone? |
| 15:46.55 | Samot | There is no Asterisk client for mobile phones. |
| 15:47.25 | Samot | You would install a softphone client that would connect to Asterisk. |
| 15:47.50 | Samot | If you want to forward calls from your cell phone to Asterisk, you need to forward calls to a real number like normal. |
| 15:48.39 | Zombie | I take it that would require a VOIP Provider? |
| 15:48.58 | Samot | Just as I said. |
| 15:49.19 | Samot | Otherwise whatever you use to connect to Astersik, be it a hardware based phone or a softphone is just connected to Asterisk. |
| 15:49.30 | Samot | They will be able to call Asterisk and each other. |
| 15:49.42 | Samot | There still needs to be a connection to the PSTN which requires a provider. |
| 15:49.55 | Samot | So you can make and receive calls from other people. |
| 15:51.02 | Zombie | Well. |
| 15:51.54 | *** part/#asterisk igcewieling (~ewieling@199.27.202.69) |
| 15:55.34 | Zombie | I'd still like to get the internal stuff. |
| 15:55.38 | Zombie | working. |
| 15:57.41 | Zombie | As a start. |
| 15:57.50 | Zombie | And see where I can take this. |
| 16:08.25 | drmessano | Step1. Ditch users.conf |
| 16:09.00 | Samot | OK, so what internal stuff do you want working? |
| 16:09.21 | drmessano | He can't yet call ext to ext |
| 16:09.30 | drmessano | The pastebin he showed last night was users.conf |
| 16:09.43 | drmessano | I told him to get a proper chan_pjsip config |
| 16:09.51 | *** join/#asterisk yuljk (~yuljk@unaffiliated/yuljk) |
| 16:09.53 | drmessano | and then hours later he pinged "anyone still awake" |
| 16:10.21 | drmessano | If someone wants to troubleshoot his users.conf, go for it, but what a waste |
| 16:11.49 | Samot | Oh I saw that hot garbage of a config |
| 16:12.33 | drmessano | I've seen this conversation repeated multiple times in my scrollback. This is turning into CV |
| 16:13.44 | Samot | https://i.imgflip.com/3wyrvt.jpg |
| 16:14.57 | Samot | Speaking of CV I saw someone on DSLReports, I think, asking "I got an old PAP2 laying around. What can I do with it?" |
| 16:15.35 | Zombie | I want to show you what version of Asterisk I have |
| 16:15.51 | Zombie | asterisk-13.7.2 |
| 16:15.57 | Samot | Wow. |
| 16:16.10 | Zombie | I have put in a ticket with My Distro to have this upgraded. |
| 16:16.29 | Samot | Which distro? |
| 16:16.32 | Zombie | That request is being processed. |
| 16:16.38 | Zombie | Rosa and Mageia. |
| 16:19.05 | drmessano | Zombie: Do you yet understand that you need chan_pjsip configured and not the useless garbage of users.conf? |
| 16:19.23 | Zombie | Yes I do. |
| 16:19.37 | Samot | has a strange urge to listen to the Cranberries.... |
| 16:20.01 | Zombie | Could it be because I am here? |
| 16:20.13 | Zombie | (and I do like Zombie by the Cranberries.) |
| 16:20.23 | drmessano | So you keep parroting that you want the basics to work.. great. Get your stuff crammed into a chan_pjsip config and go from there |
| 16:31.05 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 16:31.27 | Zombie | So this file? /etc/asterisk/pjsip.conf |
| 16:31.52 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 16:32.24 | Zombie | considers moving all files in /etc/asterisk/ to /etc/asterisk/examples |
| 16:52.31 | Zombie | https://pastebin.com/JMiDgHpb |
| 17:03.07 | Zombie | drmessano, Its more broken than it was. |
| 17:03.14 | Zombie | I can't do sip show peers |
| 17:03.57 | drmessano | sip x x is chan_sip, the prefix is pjsip x x |
| 17:05.41 | Zombie | olympia*CLI> show sip |
| 17:05.41 | Zombie | No such command 'show sip ' (type 'core show help show sip' for other possible commands) |
| 17:05.41 | Zombie | olympia*CLI> pjsip show peers |
| 17:05.41 | Zombie | No such command 'pjsip show peers' (type 'core show help pjsip show' for other possible commands) |
| 17:05.41 | Zombie | olympia*CLI> pjsip show peers |
| 17:06.33 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+chan_pjsip+CLI+Specification |
| 17:06.41 | drmessano | Dude, stop getting and look at the wiki |
| 17:06.48 | drmessano | Dude, stop guessing and look at the wiki |
| 17:06.56 | drmessano | It's all documented |
| 17:07.06 | sibiria | Zombie: if you aren't loading any modules it will be a bit complicated to get anything going |
| 17:10.24 | sibiria | in my opinion, the default config set you get from "make samples" is a good starting point |
| 17:10.30 | sibiria | don't remove this set, just alter it instead |
| 17:10.42 | Zombie | Thats what I was doing. |
| 17:14.39 | sibiria | also, pjsip show contacts |
| 17:14.45 | sibiria | and, pjsip show endpoints |
| 17:15.13 | drmessano | Zombie: I told you to dump users.conf, not dump the whole config so modules don't load |
| 17:15.54 | drmessano | There's a whole wiki full of startup information |
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| 19:51.49 | Cyrillax | Can I get SIP RealTime to use a table for device hints? Trying to get Queues() to use an externally sync'd DB for Device Hints with Local/ members, but want to keep Queue()'s ability to filter out based on the Hints |
| 19:53.37 | Cyrillax | (Without using AMI or something to maintain a full copy of hints on each *) |
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| 20:23.38 | DannyA | hey all. so my asterisk stops logging when i exit the CLI. any idea why that is? |
| 20:24.05 | DannyA | the "full" log contains all the normal entries you'd expect to see, but as soon as i exit the CLI or my SSH session ends, the logging stops |
| 20:28.07 | igcewieling | never heard of that. |
| 20:28.18 | igcewieling | how are you starting Asterisk? |
| 20:30.00 | DannyA | i guess in init.d |
| 20:30.28 | DannyA | but i confirmed it just now. i logged into the CLI and did logger reload, in a separate ssh session i cat'd the file and saw it had entries, then exited the CLI session, and logging stopped |
| 20:44.39 | igcewieling | nope, never seen that. |
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| 20:55.11 | Dovid | Is there any application like BackGround that will play a sound file, ignore the DTMF but stop the playback and keep moving in the IVR? I want it to behave like Background that it stops playing the file but then act like playback that we ignore the dtmf |
| 20:57.59 | DannyA | why not just do Background but then have a pattern match for extension for any digit? |
| 21:02.35 | Dovid | DannyA: I ended up using Read |
| 21:03.08 | Dovid | I cant because I play instructions that I want users to be able to skip and continue in the priorities. using background will jump out of it and create a loop |
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| 22:31.48 | dogcow | hello all, I am looking to buy a Digium T1/E1 card (used), and I'm trying to determine if the older PCI (not PCI-e) models will still be supported in Asterisk 16. Does anyone know? Thanks in advance. |
| 23:04.38 | joepublic | mine works with asterisk 16. |
| 23:04.59 | joepublic | ah, scratch that, I have a TDM400P |
| 23:05.56 | drmessano | dogcow: If you can only afford a used older modem, maybe get a new generic card |
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| 23:07.37 | drmessano | Like an OpenVOX for $300 |
| 23:09.24 | drmessano | Someone's used, and maybe soon to be unsupported card vs generic card that seems to work OK |
| 23:14.50 | dogcow | thanks, drmessano, I'll look at OpenVOX. Hadn't heard of them. $300 is about the top of my budget. (Just for home lab use). |
| 23:15.25 | drmessano | I used OpenVOX for my home lab use years ago. They're fine |
| 23:16.04 | drmessano | For a multi-million dollar business, Digium all the way. |
| 23:16.11 | drmessano | But home use, yeah OpenVOX |
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