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| 01:18.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.3 (2017/12/01), Standard: 15.1.3 (2017/12/01); 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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| 06:55.46 | Fesuy | can I test asterisk without hardware? |
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| 07:00.05 | [TK]D-Fender | You do need a computer to run it on. |
| 07:00.18 | [TK]D-Fender | beyond that you can test whatever aspects of it you want by whatever means you want |
| 07:02.32 | Fesuy | Thanks [TK]D-Fender. That's what I want :) |
| 07:02.56 | [TK]D-Fender | You can learn all about the dialplan, process calls, use conferencing features, etc |
| 07:03.18 | [TK]D-Fender | All you should have at a minimum is preferrably 2 "phone" type devices of any kind. |
| 07:03.59 | [TK]D-Fender | This can be soft-phones on PC's or on smart-phones so as not to nee more than basic pc / phone / tablet tech |
| 07:04.18 | [TK]D-Fender | Enough so you can place a call, process the request and dial another device and have a converstaion. |
| 07:04.25 | [TK]D-Fender | Beyond that a call is a call is a call is a call |
| 07:09.18 | Fesuy | what the best free softphone for linux? |
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| 07:11.58 | [TK]D-Fender | linphone, ekiga, etc |
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| 07:29.41 | [TK]D-Fender | Bed time, BBL |
| 07:48.12 | jkroon | are other people also noticing memory leaks in the last asterisk 11 (11.25.3) release? along with chan_iax2 problems that was not present previously? |
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| 08:07.51 | bittis | <PROTECTED> |
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| 10:13.51 | Penguin | How can I execute a script and pass args when a new voicemail is left in mailboxes? |
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| 10:31.21 | thiagoc | Hi all... I have this same issue: https://community.asterisk.org/t/not-audio-pjsip-2-5-and-asterisk-14-nat/68776 |
| 10:31.41 | thiagoc | here's the call log: https://pastebin.com/raw/QayrjEts |
| 10:31.49 | thiagoc | any suggestions? |
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| 16:42.48 | kitsune | Hey, so I have a weird problem. |
| 16:43.51 | kitsune | When I initiate a call from my phone, asterisk doesnât send any RTP packets until it receives RTP packets from the far end. This requires me to open up a lot of ports. |
| 16:44.14 | Samot | That's how RTP works. |
| 16:44.21 | kitsune | And this happens regardless of whether I use symmetric-rtp or not (pjsip). |
| 16:44.29 | Samot | Asterisk will never know where to send RTP until the other side tells it. |
| 16:44.48 | Samot | Asterisk does not know what the other end users for RTP automatically. |
| 16:44.55 | Samot | Just like the other end doesn't know what Asterisk uses. |
| 16:45.00 | Samot | Until Asterisk tells it. |
| 16:45.20 | kitsune | The SIP messages negotiate the ports correctly. |
| 16:45.57 | kitsune | However the NAT doesnât set up the pinhole into the Asterisk server until Asterisk sends traffic out of that port to that IP address. |
| 16:47.11 | Samot | Who's NAT? |
| 16:47.15 | kitsune | Even in the case of symmetric RTP, it would be useful to send preliminary RTP packets to the far endâs designated RTP port, even if that port is incorrect. |
| 16:47.25 | Samot | Is Asterisk behind NAT? |
| 16:47.30 | kitsune | Yes. |
| 16:48.03 | Samot | Until the other end sends SDP Asterisk doesn't know _where to send RTP_ |
| 16:48.13 | Samot | The IP could be different |
| 16:48.14 | kitsune | Sorry, should have clarified that. |
| 16:48.22 | Samot | The ports, for sure. |
| 16:48.30 | Samot | The RTP ports do not "listen" |
| 16:48.31 | kitsune | The SIP negotiation seems fine. |
| 16:48.35 | Samot | OK |
| 16:48.47 | kitsune | The far end begins sending a media stream |
| 16:48.51 | Samot | Asterisk will not know the RTP port of my Polycom until my Polycom tells it. |
| 16:49.08 | kitsune | The media stream hits the firewall and gets dropped. |
| 16:49.14 | Samot | Where? |
| 16:49.16 | Samot | On Asterisk? |
| 16:50.26 | file | kitsune: you'll need to be specific as to the call scenario - generally though Asterisk doesn't generate media itself unless you do something that would trigger it (playing back a file for example), in a 2 party call it forwards media |
| 16:50.53 | Samot | And which side's NAT/Firewall is doing something. |
| 16:51.01 | file | both chan_sip and chan_pjsip have an RTP keepalive option to send an RTP packet periodicially as a keep alive |
| 16:51.07 | Samot | Because both the devices and Asterisk are behind NAT. |
| 16:51.35 | kitsune | file: That is helpful maybe. |
| 16:51.55 | Samot | Which _firewall_ is dropping the traffic? |
| 16:52.03 | Samot | The one in front of Asterisk or where the devices are? |
| 16:52.37 | kitsune | The one on the Asterisk side. |
| 16:53.01 | Samot | So it's dropping incoming traffic to the IP on ports 10000-20000? |
| 16:53.26 | voipmonk | UDP ( always ) |
| 16:53.28 | Samot | Or whatever your RTP port range is set to. |
| 16:54.16 | kitsune | So if I port forward a huge range of ports to Asterisk (10000-20000) it works. But what I donât understand is why Asterisk bothers to wait until it received RTP packets to send its own. |
| 16:54.30 | Samot | Because that's how RTP works. |
| 16:54.39 | file | we actually will send |
| 16:54.42 | file | if there's any to send |
| 16:54.49 | Samot | Not all SIP transactions do not require SDP/RTP |
| 16:55.04 | Samot | If the other side sends SDP then things happen |
| 16:55.17 | Samot | It if doesn't and Asterisk starts it, it will send SDP to the other side. |
| 16:55.19 | kitsune | Okay wait now that is interesting. |
| 16:55.32 | kitsune | So maybe the far end is also getting blocked. |
| 16:55.36 | kitsune | err |
| 16:55.39 | kitsune | the carrier |
| 16:55.40 | Samot | REGISTER/SUBSCRIBE/NOTIFY/OPTIONS |
| 16:55.42 | Samot | ^^ No SDP |
| 16:56.16 | file | without an Asterisk log and "rtp set debug on" that's all I can sya |
| 16:56.28 | Samot | What type of firewalls? |
| 17:00.06 | kitsune | The rtp debug is literally nothing. I think I see whatâs going on though. |
| 17:00.36 | kitsune | The caller is being bridged with a carrier, which is obviously also outside the NAT |
| 17:00.42 | file | yup |
| 17:00.49 | file | so there's nothing for Asterisk to forward. |
| 17:01.10 | Samot | How is that outside the NAT? |
| 17:01.15 | Samot | If Asterisk is behind NAT |
| 17:01.25 | Samot | Then the carrier is going through NAT |
| 17:02.21 | kitsune | Yeah so is there anything that can be done about that to get the ball rolling? |
| 17:02.36 | Samot | We haven't seen anything from this. |
| 17:02.54 | Samot | Don't know how Asterisk is setup for the NAT/local networks/etc. |
| 17:03.09 | Samot | What type of firewall is in front of Asterisk? |
| 17:03.46 | file | kitsune: you can enable the RTP keepalive option as I said which will send a packet out at an interval, including at the start |
| 17:03.57 | file | or you can have Asterisk generate media in some form (like playing back a 1 second silence) |
| 17:04.01 | file | or you can forward ports |
| 17:04.42 | kitsune | Okay, thanks. The keepalive seems promising so Iâll start with that. |
| 17:04.50 | Samot | What type of firewall is in front of Asterisk? |
| 17:05.11 | file | Samot: hush, I know the problem and what I said are the options |
| 17:07.05 | kitsune | Itâs AWS, for the record. |
| 17:07.17 | Samot | Hah. |
| 17:07.18 | Samot | OK |
| 17:09.05 | Samot | Where do inbound calls land? |
| 17:09.24 | Samot | What do they hit? Do you just Dial() straight out to a device/endpoint? |
| 17:09.27 | Samot | Do they hit an IVR? |
| 17:13.12 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
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| 17:29.53 | *** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.4 (2017/12/13), Standard: 15.1.4 (2017/12/13); 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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| 17:50.00 | kitsune | Still working on it... The keepalive works on at least one branch very well. |
| 17:54.12 | Samot | Do they hit an IVR? |
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| 19:02.26 | Demon_VoIP | file, is there any certainty when it will be fixed ASTERISK-26806? Regular hangs and reboots very upset :( |
| 19:02.54 | file | I'm rewriting it in my spare time. |
| 19:03.05 | file | It'll be done and through review when it is done. |
| 19:03.22 | file | Probably in a few weeks. |
| 19:04.00 | Demon_VoIP | That would be great. I'm waiting for this for a year. |
| 19:05.51 | Demon_VoIP | Currently, asterisk crashes a couple of times a week. Mostly hangs, it seems that when reloaded. |
| 19:06.18 | file | seanbright: you cheeky bugger |
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| 19:21.59 | seanbright | :D |
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| 20:10.56 | maruen | Hello everybody, anybody could help to have a script that changes the IMEI from the dongles? |
| 20:11.18 | maruen | My carrier blocked all my dongles through it's IMEI |
| 20:11.24 | maruen | and I'm trying to solve that issue now |
| 20:11.27 | maruen | anyone could help on it? |
| 20:20.04 | RovingWriter | why'd they do that? |
| 20:21.00 | [TK]D-Fender | Because he's abusing their ToS |
| 20:21.52 | maruen | That ToS means? |
| 20:22.00 | [TK]D-Fender | Terms Of Service |
| 20:22.08 | [TK]D-Fender | they are blocking you for a reason |
| 20:22.09 | maruen | I'm sending SMS's to my customers |
| 20:22.19 | [TK]D-Fender | Don't expect help in trying to circumvent their rules. |
| 20:22.20 | maruen | and they don't want to allow me to do that |
| 20:22.42 | maruen | asterisk wouldn't exist then if it would follow strictly carrier rules |
| 20:22.42 | [TK]D-Fender | Go find another service that does |
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| 20:23.06 | [TK]D-Fender | * existed before you installed it, and will exist long after I'm sure as well |
| 20:23.41 | maruen | You didn't understand what I meant |
| 20:24.29 | [TK]D-Fender | Same here |
| 20:24.57 | [TK]D-Fender | If they don't want you using automated tools to send SMS then we aren't here to try to help you cheat |
| 20:25.28 | [TK]D-Fender | <[TK]D-Fender> Don't expect help in trying to circumvent their rules. <- |
| 20:28.13 | RovingWriter | [12:32] <[TK]D-Fender> <[TK]D-Fender> Don't expect help in trying to circumvent their rules. <- <- |
| 20:30.15 | Penguin | How can I execute a script and pass args when a new voicemail is left in mailboxes? |
| 20:31.18 | MrTAP | ; If you need to have an external program, i.e. /usr/bin/myapp called when a voicemail is left, delivered, or your voicemailbox is checked, uncomment this. |
| 20:31.18 | MrTAP | ;externnotify=/usr/bin/myapp |
| 20:31.33 | [TK]D-Fender | yup |
| 20:31.46 | Penguin | That takes args? |
| 20:33.15 | Penguin | Where can I find the documentation on how to use that? The comment isn't exactly helpful. |
| 20:33.33 | MrTAP | it should pass some args by default to tell you which mailbox etc |
| 20:33.36 | MrTAP | try it and find out |
| 20:34.08 | MrTAP | or search the code for externnotify in apps/app_voicemail.c to find out exactly what it does |
| 20:34.17 | [TK]D-Fender | You could also parse out what gts passed to the e-mail app in the smple config |
| 20:34.34 | [TK]D-Fender | you have access to a certain number of fields. Not sure if you can any beyond the basics or not |
| 20:34.38 | [TK]D-Fender | Should be an easy test |
| 20:36.10 | Penguin | Maybe I can hijack the email thing to send the email and run my script. The variables for the email are what I need to pass to the script anyway. |
| 20:36.59 | [TK]D-Fender | That should do it then depending on how it passes them (or you're able to parse backwards) |
| 20:38.22 | Penguin | If the externnotify thing does what I want it to do, I would prefer that mechanism. But I'm willing to look at alternatives. |
| 20:39.57 | [TK]D-Fender | Check what it gets passed exactly. It isn't too clear fromt he sample config so far as I've read. |
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| 20:52.01 | bipolar | Is there a way to reset the callnumber use on an IAX client IP address without restarting the whole system? |
| 20:52.28 | bipolar | I've got two ip's that have hit the 2048 limit and are not coming back down. It seems to be a glitch. |
| 20:53.57 | [TK]D-Fender | There shouldn't be a way to mess with an internal counter like that |
| 20:54.18 | [TK]D-Fender | So restart of * (or at least chan_iax2) would likely be the only way |
| 20:57.46 | bipolar | ok |
| 20:57.49 | bipolar | ty |
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| 21:41.49 | dan_j | Hi. When using confbridge, is there any way to normalise the volume of each caller so they match? I've got one caller dialling in via our sip trunk from our DID provider, and then two staff connecting via SIP directly from SIP phones. The sip trunk caller is really quiet. |
| 21:48.53 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 21:50.14 | rmudgett | dan_j: There is no automatic volume control. You can setup a confbridge DTMF menu to adjust volumes. You can use the VOLUME dialplan function. |
| 21:51.14 | Samot | You're noticing this because the call isn't on a phone, I'm pretty sure. |
| 21:51.28 | dan_j | It's complicated because the sip trunk caller is a customer, and getting them to dial a button to adjust their talk volume isn't that professional. |
| 21:51.37 | Samot | Where the phone's hardware/software deals with the audio. |
| 21:51.51 | Samot | What? |
| 21:51.57 | dan_j | Samot: I'm noticing this because normally there is only one caller talking to another caller, so nothing to compare it to. |
| 21:52.29 | Samot | Right. |
| 21:52.39 | Samot | So the phone is dealing with an single audio stream |
| 21:52.48 | Samot | Coming from the conf bridge |
| 21:53.01 | Samot | So it's going to adjust audio at the phone level overall for that stream |
| 21:53.09 | Samot | As when it was just a single call... |
| 21:53.20 | Samot | Hence my statement. |
| 21:53.33 | Samot | Which is meant to imply, the call is that low all the time over the trunk. |
| 21:53.35 | dan_j | Using other conferencing systems, I've never noticed volume differences. I suspect the conferencing services I've used in the past automatically adjust volume. Hence my question. |
| 21:54.04 | dan_j | This is the first time this issue has been noticed and I've been allowing conferences for about 6 months now. |
| 21:54.33 | dan_j | My guess is that the sip trunk (PSTN) caller didn't have a very good phone. |
| 21:54.46 | dan_j | But it would have been nice if asterisk could normalise the volume. |
| 21:54.58 | Samot | To what? |
| 21:55.01 | Samot | Normalize it to what? |
| 21:55.08 | Samot | The phones are doing that |
| 21:56.12 | dan_j | Normalise Caller A (from PSTN over SIP Trunk), Caller B (a member of staff using a SIP phone) and Caller C (another member of staff using a SIP phone), so when the audio is mixed together, all the callers are the same volume. |
| 21:56.42 | dan_j | Currently, Caller C is hearing a quiet Caller A, and a normal sounding Caller B |
| 21:56.57 | Samot | And caller B? |
| 21:57.15 | dan_j | Caller B is hearing a quiet Caller A, and a normal sounding Caller C |
| 21:57.26 | Samot | And what happens when you are Caller A? |
| 21:57.40 | Samot | As in, have you jumped into a conf bridge to replicate this with them? |
| 21:58.38 | Samot | If you call in over the PSTN via your cell phone and do this.. |
| 21:58.46 | Samot | Are they hearing you "low" or "normal" |
| 21:58.57 | dan_j | I dont know what Caller A's experience is, but my assumption is that since B and C have an identical setup, Caller A hears them equally well. |
| 21:59.06 | Samot | Wrong. |
| 21:59.11 | Samot | What happens when YOU do it? |
| 21:59.20 | Samot | And YOU are Caller A in the scenario? |
| 21:59.33 | Samot | You need to replicate this. |
| 21:59.51 | dan_j | During testing and implementation 6 months ago, I could hear fine when I dialled into the conference via PSTN |
| 21:59.56 | Samot | There are numerous factors that could be involved and some of them are out of your control. |
| 22:00.00 | Samot | Again |
| 22:00.01 | Samot | Now |
| 22:00.04 | Samot | Not 6 months ago |
| 22:00.08 | Samot | The issue is happening now |
| 22:00.11 | Samot | Have you replicated it? |
| 22:00.37 | Samot | Not only to get captures and an idea of what is happening |
| 22:00.41 | Samot | But to see if it happens to YOU |
| 22:00.50 | Samot | Because this could be an issue with the caller |
| 22:01.59 | dan_j | It won't make any different if I experience the issue or not. |
| 22:02.26 | dan_j | If it's fine for me, it means that Caller A had a bad phone or similar issue, but the volume would still need boosting in order to get it to work for him. |
| 22:02.57 | Samot | Yes. |
| 22:03.01 | Samot | It makes all the difference. |
| 22:03.10 | Samot | It is it ONE caller out of ALL callers.. |
| 22:03.13 | Samot | This isn't a system issue |
| 22:03.17 | Samot | This isn't an Asterisk issue |
| 22:03.21 | Samot | This is an issue with that caller |
| 22:03.29 | Samot | An isolated case. |
| 22:03.42 | Samot | That shouldn't require you to have to redo your configurations overall |
| 22:03.48 | Samot | or anything. |
| 22:04.04 | dan_j | Agreed, but I still need to look at this isolated case to see if anything can be improved for the future. Such as automatic normalisation of the volume received from each participant. |
| 22:04.06 | Samot | Except help the guy understand the user menu and how to control the volume. |
| 22:04.17 | Samot | Which, by the way, is part of what some conference services do |
| 22:04.25 | Samot | They give you a "Here's your options" |
| 22:04.34 | dan_j | Ok. i hear |
| 22:05.03 | dan_j | I'll inform Caller's B and C about the options so they can inform Caller A if it occurs again. |
| 22:05.25 | Samot | And this is the only caller they are reporting this for? |
| 22:05.31 | Samot | This one particular caller? |
| 22:06.24 | dan_j | Honestly, Caller B reporting the issue is a new member of staff. it may be that the other staff members have experienced this but considered it normal. I have asked the client to confirm whether it's a one-off case or part of a bigger issue. |
| 22:06.51 | dan_j | just wanted to know if there was a way to automate the whole thing. Like normalize_volume=yes :) |
| 22:07.18 | rmudgett | Nope. No automatic gain control. |
| 22:10.38 | dan_j | Np. Thanks. |
| 22:10.54 | Samot | And there is two volume controls. |
| 22:11.00 | Samot | Talking and Listening |
| 22:11.18 | dan_j | Yep. Is it possible for Caller B to control Caller A's talk volume? |
| 22:11.38 | dan_j | Correction... is it possible for Caller B and C to control Caller A's talk volume? |
| 22:11.55 | dan_j | Or only Caller A can control his talk volume? |
| 22:11.59 | Samot | No |
| 22:12.08 | Samot | You control your own channel. |
| 22:12.22 | Samot | Caller can turn their listening/talking volume up or down |
| 22:13.18 | dan_j | Ok. Shame. Alright, i'll let B and C know to tell A how to increase the talk volume |
| 22:16.44 | *** part/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 22:26.17 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:28.46 | [TK]D-Fender | dan_j, Or go set up a dynamic feature to up it |
| 22:58.17 | *** join/#asterisk z3ntu (z3ntumatri@gateway/shell/matrix.org/x-fplcodioenjmojqv) |
| 23:16.59 | *** join/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 23:19.07 | SwK | anyone alive? |
| 23:19.23 | [TK]D-Fender | nope, all dead |
| 23:19.58 | SwK | heh |
| 23:20.03 | SwK | whats up dude LTNS |
| 23:23.17 | SwK | so someone refresh my memory... doesnt this log line mean asterisk couldnt find the user so it never even tried to send the invite? (like user not registered or something) |
| 23:23.19 | SwK | app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 23:23.45 | [TK]D-Fender | Either no IP to contact them, or they are considered unreachable |
| 23:24.38 | SwK | thats what I thought been living in the land of the other software for a decade |