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| 00:37.08 | wonderworld | hi, i have a question regarding dtls_rekey. do i need to provide my own certificate when using rekeying as some kind of "seed" or will asterisk generate all needed keys by itself when using rekeying? |
| 00:37.17 | wonderworld | (in pjsip) |
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| 02:19.06 | velix | Is there a way to set pjsip history "always on" ? |
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| 02:31.36 | Samot | The logger history? |
| 02:54.20 | igcewieling | see cli.conf ? |
| 02:54.29 | igcewieling | see pjsip.conf |
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| 03:01.33 | *** join/#asterisk Tashtari (~tashtari@unaffiliated/tashtari) |
| 03:04.21 | Tashtari | This might not be exactly the right place to ask this, but hopefully someone here can at least point me in the right direction. I'm looking to interface an old-fashioned two-wire door intercom as an IP phone endpoint - i.e. someone outside the door presses the call button and it initiates a call to a fixed number, and some bit of electrical magic handles the switching of the intercom device between being a |
| 03:04.23 | Tashtari | speaker and a microphone. |
| 03:04.34 | Tashtari | Any idea what I might be looking for? |
| 03:11.57 | drmessano | Well |
| 03:12.08 | drmessano | an ATA, set to auto-dial would work |
| 03:12.37 | drmessano | Not sure what impedence the intercom presents |
| 03:13.04 | Tashtari | What's an ATA? |
| 03:13.23 | Tashtari | For what it's worth, I think the answer to that is 20 ohms. |
| 03:13.28 | Tashtari | (The impedance) |
| 03:13.36 | drmessano | Analog telephony adapter for converting an analog line to SIP |
| 03:13.58 | drmessano | Is there a seperate mic and speaker? |
| 03:15.21 | Tashtari | Nope, they're one and the same. |
| 03:16.32 | Tashtari | The thing just interfaces with two wires; it seems that pressing the button shorts them together through a 3.7 ohm resistor. |
| 03:17.22 | drmessano | huh ok |
| 03:17.43 | Samot | For the most part, the ATA suggestion is the best one. |
| 03:18.42 | drmessano | So you would need low impedence to 600 transformer |
| 03:19.56 | Tashtari | hm. I should say that my knowledge of electrical engineering is pretty scant. Is this too esoteric a use case to have an off-the-shelf part? |
| 03:20.13 | drmessano | Oh you won't do this with anything off the shelf |
| 03:21.02 | Tashtari | Darn. |
| 03:21.33 | drmessano | You need to build an interface, not just to match impedence, but also to hold/hang up the line |
| 03:22.25 | Tashtari | Well, that will be interesting. Am I right to assume that basically any way I do this will involve effectively converting this thing into an analog telephone and then plugging it into an ATA? |
| 03:22.42 | drmessano | More or less, yes |
| 03:22.46 | Tashtari | Hmm. |
| 03:22.55 | Tashtari | All right, that should make for an interesting challenge... |
| 03:23.04 | Tashtari | All righty, thanks for the pointer. |
| 03:23.15 | drmessano | Sure thing |
| 03:23.43 | *** part/#asterisk Tashtari (~tashtari@unaffiliated/tashtari) |
| 03:23.57 | Samot | OK then. |
| 03:24.21 | igcewieling | http://help.nyigc.net/doc/Algo_SIP_Endpoints.pdf |
| 03:25.03 | drmessano | Nothing there that would help them |
| 03:25.47 | drmessano | "I'm looking to interface an old-fashioned two-wire door intercom as an IP phone endpoint" |
| 03:27.03 | Samot | Speaking of old equipment..I wonder if that one guy ever got his headmaster lamp working. |
| 03:27.18 | drmessano | Oh yeah |
| 03:28.16 | Samot | He seemed to have most of it down. |
| 03:28.37 | igcewieling | perhaps a sip relay controller https://www.algosolutions.com/product/8061-ip-relay-controller/?v=7516fd43adaa |
| 03:28.55 | igcewieling | Algol has all sorts of cool stuff. |
| 03:30.05 | Samot | drmessano: Darn, thread is closed. Looks like he was messing around with just the AMI part but had the rest sorted. "MQTT message to the ESP32 which in turns triggers the relay" |
| 03:32.12 | drmessano | You'd have to build almost as much for that relay controller as you would an ATA |
| 03:32.34 | Samot | Yeah, that's why it came to mind with this. |
| 03:32.36 | drmessano | I wonder if the guts from an old telephone would work |
| 03:33.13 | Samot | "It's got ATT guts. This is the one you want" |
| 03:33.23 | Samot | But it might. |
| 03:34.07 | Samot | A good ole 1970's era table phone. One you could commit murder with. |
| 03:34.52 | Samot | Western Electronic keeping you connected and murdering since 1960. |
| 03:35.53 | drmessano | Even an AT&T 210 could work. I seriously doubt they changed the design to where the mic and speaker were that far apart. |
| 03:37.34 | *** join/#asterisk cyford (~cyford@65.254.218.68) |
| 03:48.19 | velix | igcewieling: Hmm, cannot find anything in pjsip.conf. Does cli.conf start the history only when starting the cli or does it start when starting asterisk (I'm running it as daemon)? |
| 03:48.30 | velix | don't answer |
| 03:48.45 | velix | "Any commands listed in this section will get automatically executed when Asterisk starts as a daemon or foreground process (-c)." |
| 03:59.08 | Samot | velix: Are you asking about the pjsip logger history? |
| 03:59.23 | velix | Samot: not the logger, just the history. |
| 03:59.35 | velix | But igcewieling gave me the right hint. |
| 03:59.46 | velix | [startup_commands] |
| 03:59.46 | velix | pjsip set history on = yes |
| 04:02.47 | Samot | It will eat up memory |
| 04:02.54 | Samot | So you shouldn't really leave it on all the time. |
| 04:07.00 | velix | Samot: no, of course not. |
| 04:07.06 | velix | Just for debuuging right now. |
| 04:07.12 | velix | But memory is my smallest problem ;) |
| 04:07.24 | Samot | Then you don't need to make it a start up option. |
| 04:07.31 | Samot | pjsip set history on |
| 04:07.32 | velix | I often forget to turn it on. |
| 04:07.38 | velix | And then after hours, I'm sad. |
| 04:07.51 | Samot | It's not meant as a logging tool to look at later. |
| 04:08.18 | velix | Before having an ip filter, I had an 1 GB security log file after a half day :) |
| 04:08.40 | velix | After having an IP filter (whitelist German IP ranges only), it dropped to nearly zero |
| 04:08.58 | Samot | Uhm. |
| 04:08.59 | velix | Samot: Okay, I'll turn it off again. |
| 04:09.16 | Samot | I didn't say it was going to eat up disk space. I said it was going to eat up memory. |
| 04:09.29 | Samot | It stores all that information _in-memory_ |
| 04:09.33 | velix | Yeah, but RAM isn't a problem here ;) |
| 04:09.34 | Samot | Not in a log file. |
| 04:09.41 | Samot | OK. |
| 04:09.53 | velix | This server has 12 cores and 64 GiB RAM. |
| 04:09.57 | Samot | OK. |
| 04:10.38 | velix | (actually the CPU has 24 cores + threads, but I've got 12 vCores only) |
| 04:10.42 | velix | vCPUs |
| 04:11.02 | Samot | How many PJSIP endpoints are there? |
| 04:11.09 | Samot | How many calls per day does this system make? |
| 04:11.19 | velix | max. 10 |
| 04:11.26 | Samot | Per day? |
| 04:11.26 | velix | 10 endpoints in parallel (10 students) |
| 04:11.30 | Samot | OK. |
| 04:11.31 | velix | and calls... about 140-170 |
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| 04:11.43 | Samot | So 10 endpoints that are registering and have keepalives. |
| 04:12.00 | velix | Yeah, but the machine was a spare machine for data mining and is way too big to this job. |
| 04:12.11 | Samot | Alright. |
| 04:12.28 | Samot | Seems you got it cover. Add it as a start up function. |
| 04:12.35 | Samot | Let it keep filling up memory. |
| 04:12.46 | Samot | Eventually you'll stop having working calls |
| 04:13.20 | velix | hehe, mate, I _do_ have stop having working calls ;) |
| 04:13.24 | velix | That's why I'm logging :D |
| 04:13.39 | Samot | Here's the answer. |
| 04:13.40 | velix | But I know what you mean ;) |
| 04:13.50 | Samot | Dump the University SIP trunk. It's hot garbage. |
| 04:14.06 | Samot | They are not very capable and it is the key source to all your issues. |
| 04:14.06 | velix | Yep. The commercial one is working without a problem. But calls to mobile phones are damn expensive. |
| 04:14.21 | Samot | You realize that fixing this problem constantly is a cost too |
| 04:14.31 | velix | True, true :( |
| 04:14.32 | Samot | Well I guess when you work for free, what does it matter. |
| 04:14.57 | velix | Sure, it's 6 am ... I'm here since 12 hours. I don't get paid for this. |
| 04:15.08 | velix | But I'm learning Asterisk. So I could help others one day. |
| 04:15.26 | Samot | OK.. |
| 04:15.47 | Samot | Look, it's one thing to sit here and say "I'm learning Asterisk" when it's full of problems from Asterisk. |
| 04:16.01 | Samot | But when the issue is 100% the provider's problem, you're not learning Asterisk. |
| 04:17.15 | velix | :/ you're right by 100% |
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| 09:13.45 | slima | Hi, I trying upgrade dialplan from Asterisk 11 to 16, And have a truble with setting 'accountcode' in CDR record (MYSQL), for incoming call. I have h exten, and try: h,n,Set(CHANNEL(accountcode)=${CDR(dst):2}) but in the CDR record accountcode is empty. When i set: Set(CDR(accountcode)=${CDR(dst):2}) there is a deprecated warning, and accountcode is empty in the record. How to set accountcode in h exten? |
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| 11:34.02 | *** 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 |
| 11:44.14 | *** join/#asterisk khaled-atteya (~khaled_at@197.62.157.45) |
| 11:48.11 | khaled-atteya | Hi , I use Centos 7 + asterisk 16.9.0 + dahdi 2.11.1+2.11.1 + TDM400P with 2 FXO + 2 FXS , when I add "fxsks= 1" to /etc/dahdi/system.conf the following error appear : https://paste.centos.org/view/raw/1d90d12e , what may the problem ? |
| 12:01.59 | electronic_eel | khaled-atteya: I have a very similar setup working |
| 12:02.46 | electronic_eel | with the "fxsks= 1" (or "fxoks") parameter, you tell the dahdi driver which socket on the TDM-card has which kind of interface |
| 12:02.58 | electronic_eel | the number is the socket number on the TDM card |
| 12:03.59 | electronic_eel | so maybe one of your FXO modules is in slot 1 and the FXS module you want to configure in some other slot |
| 12:04.14 | electronic_eel | so check your TDM card |
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| 13:03.06 | khaled-atteya | electronic_eel, I have 2 FXO & 2 FXS , the FXO ports are in port 1 & 2 , so I put fxsks = 1,2 |
| 13:03.23 | khaled-atteya | and the same issue with FXS ports |
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| 13:13.19 | peetaur | Can someone please help me understand why a dialplan include isn't working as expected? Just 3 lines in order works, but if I move the first to an include, it doesn't happen before the call happens. Here's the code https://bpaste.net/RHJA |
| 13:14.11 | Samot | Because the original context had a match. |
| 13:14.36 | peetaur | so how do I make it match the include first (so the first can be generated, and the custom one hand written) |
| 13:14.43 | Samot | It doesn't matter if the include => is first or last. If the current context has a match, it won't look in the includes. |
| 13:15.54 | peetaur | if _. matches everything, what can I control from the include? |
| 13:16.19 | Samot | Well I have no idea what you're trying to match or so.... |
| 13:16.32 | peetaur | I want to match everything, but do that action before calls |
| 13:16.38 | Samot | But if you want the includes to be used then there can't be a match in the original context. |
| 13:16.48 | Samot | Then you have to match and jump to each context. |
| 13:17.14 | peetaur | so I have to control where to jump before I get to the one with the include? |
| 13:17.37 | sibiria | the first thing i see is that you have two prio 1 |
| 13:17.48 | Samot | Again, if there is a match in the original context, the include won't get called. |
| 13:17.52 | sibiria | from-trunk-iax2-bc-peer-custom has one entry for prio 1 |
| 13:18.00 | sibiria | and so does the context including that one |
| 13:18.14 | peetaur | the generated one (from FreePBX) set its own priority... I would rather it was 2 so I can set lower, but can't as far as I know |
| 13:18.24 | Samot | You can't. |
| 13:18.25 | sibiria | 1,Playback(...) 1,Set(GROUP()...) |
| 13:18.38 | Samot | He's trying to use pregenerated dialplan from FreePBX |
| 13:18.50 | sibiria | so the latter instance of prio 1 is being overwritten, hence no call to Playback() |
| 13:19.56 | Samot | from-trunk-iax2-bc-peer-custom will never be called from from-trunk-iax2-bc-peer |
| 13:20.01 | Samot | Because from-trunk-iax2-bc-peer MATCHES |
| 13:20.35 | Samot | It will set the Group and then go to from-trunk |
| 13:20.41 | Samot | It will never look at from-trunk-iax2-bc-peer-custom |
| 13:22.21 | sibiria | unless my memory is off, includes in the dial plan are done when the dial plan is loaded, not when called |
| 13:22.45 | sibiria | i.e. if ,1,Set(GROUP()...) is changed to prio n, that context should run fine |
| 13:22.57 | sibiria | from-trunk-iax2-bc-peer that is |
| 13:23.39 | peetaur | it won't let me change the 1 to n... it says something like "next can't be the first priority" |
| 13:23.45 | sibiria | aha, ok |
| 13:23.49 | sibiria | yeah that sucks :) |
| 13:23.53 | peetaur | (in warnings from "core reload" on console) |
| 13:24.04 | Samot | "The order in which Asterisk tries to find a matching extension is always current context first, then all the include statements." |
| 13:24.22 | peetaur | yeah I wish it was more procedural code ..performance be damned...CPU is cheap |
| 13:24.33 | sibiria | well, there is AEL |
| 13:24.57 | sibiria | lua dial plan config |
| 13:25.04 | peetaur | ooh that sounds good |
| 13:25.19 | peetaur | but... if FreePBX is not using it, will my custom one be able to do something? |
| 13:25.22 | Samot | OK stop. |
| 13:25.32 | Samot | It is now clear that FreePBX is the wrong choice. |
| 13:25.36 | Samot | So get rid of it and start over. |
| 13:26.29 | peetaur | that's an option, but seems unnecessary if I can still hook into it...or even patch it; I like that it does 90% of what I want with very little setup |
| 13:26.49 | *** join/#asterisk Mugge (~andy@151.67.22.215) |
| 13:27.00 | Samot | Then you need to spend more time learning it. |
| 13:27.09 | peetaur | yep, and that's what I hope to accomplish today. |
| 13:27.15 | Samot | Hah. |
| 13:27.16 | Samot | OK. |
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| 15:13.09 | igcewieling | How would I confirm direct media is used? |
| 15:13.54 | igcewieling | In the Asterisk CLI if possible. |
| 15:15.57 | *** join/#asterisk irrgit (~ch33se@192.241.175.183) |
| 15:24.30 | Samot | Are you transcoding? |
| 15:25.06 | igcewieling | no. |
| 15:25.43 | Samot | Well as long as its the same codec on both sides and not transcoding, Asterisk should just user direct media. Unless it is disabled. |
| 15:25.58 | Samot | use* |
| 15:26.00 | igcewieling | rtp set debug on indicates that might not be the case. |
| 15:26.11 | igcewieling | Looks like the answer is "you can't" |
| 15:26.18 | *** join/#asterisk wk057 (~wk@unaffiliated/wizkid057) |
| 15:27.02 | wk057 | hi. is there any way to override DND set on a phone, mainly for paging? something similar to SIPAddHeader(Alert-Info: info=alert-autoanswer), but still work if the phone is on DND? |
| 15:27.22 | wk057 | (cheap cisco desk phones) |
| 15:27.22 | Samot | wk057: That is completely a phone specific feature. |
| 15:27.35 | peetaur | is the question just about whether you can detect whether it's transcoding? What I did to do that is just ban everything except codecs I expected/wanted, and it still worked...ban what I expected, and it didn't work. |
| 15:27.38 | wk057 | I kind of figured |
| 15:28.09 | wk057 | I was just hoping to find a way for a group page thing to work even if a phone was on DND |
| 15:28.26 | seanbright | depends on the phone |
| 15:28.48 | wk057 | cheap cisco SPA303 |
| 15:28.54 | wk057 | have like 8 of them in my shop |
| 15:28.57 | igcewieling | peetaur: did you use rtp set debug on to confirm media is no longer flowing via Asterisk? |
| 15:29.18 | seanbright | you can mixmonitor the call which should bring the call back in to asterisk |
| 15:29.26 | seanbright | bring the media* |
| 15:29.38 | igcewieling | I don't want media on Asterisk is possible. |
| 15:29.45 | igcewieling | s/is/if/ |
| 15:29.53 | seanbright | i'm saying as a test |
| 15:29.54 | peetaur | what I was doing was just to troubleshoot some strange phones that weren't working... I didn't need to do much debugging, just figure out which codec wasn't working. (problem was just that one of the supported codecs didn't actually work, and nothing to do with transcoding) |
| 15:30.24 | igcewieling | peetaur: do you really have no idea if direct media is in use or not. |
| 15:30.25 | seanbright | you not seeing the RTP means that it's either not flowing, or it's direct media |
| 15:30.33 | igcewieling | right. |
| 15:30.35 | seanbright | not sure that is helpful |
| 15:30.46 | peetaur | I don't know what "direct media" means ... from the above, I guessed and asked if you mean "not transcoding" |
| 15:31.12 | seanbright | direct media is when RTP flows directly between two endpoints. it doesn't go through asterisk "in the middle." |
| 15:31.37 | peetaur | ah ok |
| 15:31.49 | igcewieling | seanbright: I want to confirm direct media is in use for specific calls. One way I can do that is by turning on rtp debug, but that is hard to parse and I was hoping Asterisk would print a message. |
| 15:32.02 | seanbright | maybe i am confused |
| 15:32.14 | seanbright | if direct media is used, rtp debug should show you nothing, correct? |
| 15:32.21 | igcewieling | CORRECT! |
| 15:32.24 | seanbright | ok |
| 15:32.31 | seanbright | so there is nothing to parse |
| 15:32.33 | igcewieling | But in my case I see RTP data so obviously direct media is not working. |
| 15:32.33 | peetaur | you could also use tcpdump to look for your nothing vs something |
| 15:32.36 | seanbright | so there is nothing that is hard to parse |
| 15:32.42 | seanbright | ahh |
| 15:32.52 | seanbright | ok, so to answer your original question... |
| 15:33.03 | seanbright | if you see RTP in 'rtp debug' that means direct media is not being used |
| 15:33.07 | seanbright | i accept paypal |
| 15:33.26 | igcewieling | sends seanbright a slightly stale bagel. |
| 15:33.46 | seanbright | nothing better than a day old bagel |
| 15:34.02 | igcewieling | I suppose I can write a script to parse out the RTP debug, then use pjsip channelstats to match the rtp data to an actual channel. |
| 15:34.54 | igcewieling | I think the few milliseconds of lower audio latency might not be worth it. |
| 15:36.44 | seanbright | well. the less packet pushing asterisk is doing the better. |
| 15:38.41 | *** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@gateway/tor-sasl/ai9zo5ap) |
| 15:44.44 | igcewieling | One of the nice things about the setup on my core asterisk boxes is that there are almost no devices behind NAT |
| 15:45.16 | *** join/#asterisk miralin (~Thunderbi@195.209.246.194) |
| 15:56.18 | *** join/#asterisk CatCow97 (~mine9@c-73-96-109-206.hsd1.or.comcast.net) |
| 16:52.12 | hardwire | I can't for the life of me get res_hep to work with pjsip. Anybody have experience with this on Asterisk 16? |
| 16:53.04 | igcewieling | hardwire: installing Homer? |
| 16:53.16 | *** join/#asterisk yang (~yang@freenode/sponsor/fsf.member.yang) |
| 16:53.29 | hardwire | Right now just simple port listener.. but there is no TCP connection being initiated. |
| 16:54.34 | igcewieling | I thought HEP ran over UDP, but I'm not all that familiar with it. |
| 16:54.38 | *** join/#asterisk tmoore (~tmoore@76-231-140-59.lightspeed.sgnwmi.sbcglobal.net) |
| 16:54.52 | hardwire | makes a derp noise |
| 16:54.54 | hardwire | Thanks. |
| 16:54.57 | *** join/#asterisk atotclic (~Atotclic@unaffiliated/atotclic) |
| 16:56.18 | hardwire | what do you know.. traffic |
| 17:02.52 | hardwire | migrating from chan_sip to chan_pjsip. Anybody have a good example of a simple outbound trunk without authentication or registration that dials like PJSIP/Outbound-Trunk/${exten} ? |
| 17:03.13 | hardwire | I'm able to make what I have work by specifying a full SIP url vs ${exten} |
| 17:03.23 | file | it's PJSIP/${EXTEN}@Outbound-Trunk |
| 17:03.48 | hardwire | And then outbound server selection is done via the identify section? |
| 17:03.56 | file | identify is for inbound |
| 17:04.08 | file | outbound server selection is chosen by the configured AOR, which has a contact on it |
| 17:04.32 | hardwire | bless you |
| 17:04.39 | file | https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard |
| 17:06.14 | hardwire | Yup I reviewed the heck out of those |
| 17:06.52 | file | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 17:07.42 | hardwire | Ditto |
| 17:07.44 | hardwire | Thanks Josh |
| 17:07.57 | hardwire | Just tired from too much work and wasn't grokking. |
| 17:08.07 | hardwire | makes a hep listener and moves on |
| 17:08.43 | *** join/#asterisk _pepo_ (~Pepo@200.55.237.6) |
| 17:09.45 | _pepo_ | Hi friends, how do I change "user part" in OPTIONS message? I mean, my OPTIONS message say |
| 17:09.45 | _pepo_ | <PROTECTED> |
| 17:09.45 | _pepo_ | and I want to change to something like: |
| 17:09.45 | _pepo_ | <PROTECTED> |
| 17:21.55 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:27.23 | *** join/#asterisk miralin (~Thunderbi@5.138.77.84) |
| 17:31.59 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 17:34.31 | *** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@gateway/tor-sasl/ai9zo5ap) |
| 17:47.49 | _pepo_ | In sip.conf if I use fromuser=myname then I have OPTIONS message like |
| 17:47.49 | _pepo_ | <PROTECTED> |
| 17:47.49 | _pepo_ | How do I can change to From: "myname" <sip:myname@IPaddress> ? |
| 17:50.27 | Samot | Why does it matter? |
| 17:50.39 | Samot | The From Name is generally ignored for OPTIONs. |
| 17:51.45 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 17:52.09 | sibiria | _pepo_: with mwi_from if i recall correctly |
| 17:53.46 | Samot | Not the right option. |
| 17:55.20 | Samot | While it will set the name it will also set the user to that and it's only for MWI NOTIFYs. |
| 17:55.25 | Samot | Not going to impact OPTIONS |
| 17:55.27 | *** join/#asterisk joepublic (~joepublic@fsf/member/joepublic) |
| 18:02.57 | igcewieling | I thought mwi used SUBSCRIBE, not OPTIONS. |
| 18:04.59 | _pepo_ | I want to hide my asterisk identity |
| 18:05.11 | _pepo_ | mwi use NOTIFY (I think) |