| 00:05.18 | Labyrinth00 | Ok I will give that a try thanks |
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| 00:20.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 01:30.07 | mknooihuisen | Hi all, I wonder if someone would be able to help me out? I've got a DID from Flowroute and set it up with Asterisk 13 according to their help articles. When I dial into the DID externally asterisk claims that There is no extension MY_DID in context default. I know the default context is bad to use, but fail to see how to set the context for flowroute inbound. My sip.conf is: https://pastebin.com/zvbpj |
| 01:30.07 | mknooihuisen | 0Pb (with some obvious security blurring) |
| 01:30.28 | mknooihuisen | And that link not cut off is: https://pastebin.com/zvbpj0Pb |
| 01:35.58 | Samot | context=from-trunk <-- Does the number exist in that context? |
| 01:36.39 | Samot | Also, in their instructions "MY_DID" is a placeholder. As in, the DID you want to route. |
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| 01:51.44 | mknooihuisen | Samot: "from-trunk" is not currently a context, but I tried that. And and I know about MY_DID. I'm just using it as a placeholder here too. :) |
| 01:51.56 | Samot | OK |
| 01:51.59 | Samot | So that's your problem. |
| 01:52.12 | Samot | You're sending inbound calls to a context that doesn't exist. |
| 01:55.05 | Samot | sipdebug=yes <-- There is absolutely no need for this to be yes. You can turn sip debug on when needed. |
| 01:55.12 | Samot | Doesn't need to be on all the time, that's just nuts. |
| 01:55.48 | Samot | preferred_codec_only=yes <- Not needed if you're using just one codec. It *is* the preferred. |
| 01:56.18 | Samot | canreinvite=no <-- This should be: directmedia=no |
| 01:57.33 | mknooihuisen | Interesting, alright. I based my confs off of Flowroutes suggestions, but for clarification, I tried the [from-trunk] context and it did not work. My Extensions.conf is: https://pastebin.com/T6M6aBzW |
| 01:58.06 | mknooihuisen | I'll leave sipdebug until I get it working, then turn it off. |
| 01:58.15 | Samot | Again |
| 01:58.23 | Samot | You can just turn it on in the clie if you need to look at it. |
| 01:58.52 | Samot | OK so show an incoming call to that DID |
| 01:59.02 | Samot | asterisk -rvvvvvvvvvvv |
| 01:59.16 | Samot | or core set verbose 10 if you're already in the cli. |
| 02:00.31 | mknooihuisen | I'll assume you want sipdebug off to reduce output. |
| 02:03.25 | mknooihuisen | Samot: https://pastebin.com/7QbG08vi |
| 02:04.15 | Samot | Where do you have the default context defined? |
| 02:05.44 | mknooihuisen | To my knowledge, I don't. Although when I do define it, it works. |
| 02:06.49 | mknooihuisen | If I changed from-trunk to default in extensions.conf it would work, but I recall an astricon talk that indicated default context was bad, and also... I just want to know why it's doing that. |
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| 02:10.50 | Samot | [flowroute4][flowroute-trunk] <-- Oh this is wrong |
| 02:11.12 | Samot | Well wait... |
| 02:11.33 | Samot | Jesus, this is the one thing I hate about Flowroute. Their Asterisk stuff is always so out of date. |
| 02:11.56 | mknooihuisen | This is the guide I followed, just in case it helps: https://support.flowroute.com/SIP_Trunking_and_Voice/PBX__Configuration_Guides/Asterisk/Configure_Asterisk_13 |
| 02:12.28 | Samot | No, I know. |
| 02:12.29 | Samot | It's wrong. |
| 02:12.36 | Samot | Templates don't use brackets |
| 02:13.02 | Samot | They use parentheses |
| 02:13.18 | mknooihuisen | Okay... forgive me, been out of the loop since ARI was new. |
| 02:13.32 | Samot | That's not your fault. |
| 02:13.38 | Samot | Flowroute has the wrong instructions. |
| 02:14.05 | Samot | <PROTECTED> |
| 02:14.17 | Samot | And so do the others. |
| 02:14.25 | Samot | That's your issue |
| 02:14.46 | Samot | It's not loading the proper template with the context=from-trunk |
| 02:15.21 | Samot | So basically it sees [flowroute4] as a peer with one setting, the host |
| 02:15.29 | Samot | No context so it defaults to "default" |
| 02:16.40 | Samot | So in your sip.conf, anywhere you have [flowroute-trunk] next to another section name. |
| 02:16.56 | Samot | flowroute0, flowroute1, etc. Change the brackets to parentheses. |
| 02:17.23 | mknooihuisen | And we're happily saying hello to the world |
| 02:17.57 | Samot | https://wiki.asterisk.org/wiki/display/AST/Using+Templates |
| 02:18.32 | Samot | For the record, templates have been around for a long time. |
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| 02:18.47 | Samot | Just that Flowroute is really wrong in their example config. |
| 02:19.52 | Samot | Which is kind of funny considering that right below that derped config example it says "It becomes simple to use the template by then putting the same in () after the objects that need to use the template. " |
| 02:20.25 | Samot | ^^ That is correct. The example they show is completely wrong. |
| 02:20.43 | Samot | Someone forgot to hit the shift key when making that template. |
| 02:20.58 | mknooihuisen | Yeah... also, note that the 'register => ...' line in their code is technically in a comment |
| 02:21.12 | Samot | Well that's because it's optional |
| 02:21.21 | Samot | I don't use SIP registration with Flowroute. |
| 02:21.25 | Samot | I do IP peering/auth |
| 02:21.39 | Samot | So that line is pointless for me. |
| 02:22.05 | mknooihuisen | Ah, it's actually pointless for me too, then. It was just an attempt to fix this problem |
| 02:22.18 | mknooihuisen | I've got an IP Whitelist as well. |
| 02:22.43 | Samot | The type PEER doesn't do registration |
| 02:22.55 | Samot | Since the host is the IP/hostname you're connecting to |
| 02:23.20 | Samot | So if a call comes in from another IP/hostname, it doesn't auth on that peer. |
| 02:23.51 | mknooihuisen | Sensible enough |
| 02:26.46 | mknooihuisen | I think this answers another question I had, which is how to manage DID's from 2 separate accounts, but it looks like I can just substitute tech prefix and passwords and use other contexts |
| 02:30.37 | mknooihuisen | Anyways, thanks so much for the help, Samot. I would've been banging my head against the keyboard forever on that one. |
| 02:31.56 | mknooihuisen | As you can imagine, google didn't have any results where the default context was used, even though a different one was (supposedly) being set. |
| 02:38.34 | mknooihuisen | I just submitted a ticket to Flowroute to hopefully get this updated. |
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| 08:41.48 | Bounceman | Hi, we have a central asterisk in our environment which routes call to different upstream operators(in conjunction with Kamailio). Our upstream providers seem to generate ringtones for us, which is fine. Except for the fact that the proper indication is missing for international destinations. Is it a bad idea for asterisk (B2BUA) to generate the ringtones by its own with the proper tonezone? |
| 08:42.14 | Bounceman | Upstream <-> Kamailio <-> Asterisk <-> Kamailio <-> Customer PBX |
| 08:42.39 | Bounceman | Asterisk serves as the media gateway between the two legs |
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| 10:13.45 | mehale | hello, I need to run g729 on a virtual cloud server (Xeon (Skylake, IBRS)). I compiled it, but it causes huge delay, around 2 seconds, when I have [pstn trunk g729]<-asterisk->[pstn trunk g729] call. |
| 10:14.03 | mehale | would you suggest any already compiled version, or changing any compile option? |
| 10:14.19 | mehale | or even any asterisk setting that could be introducing this extra delay? |
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| 11:35.20 | dritskalle | i dont know anything about g729 use - but i doubt how it is compiled will cause a 2 second delay |
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| 11:41.46 | sibiria | the problem is likely in the other trunk stumbling over something |
| 11:42.48 | sibiria | i've never had any issues with delays when using g729 |
| 11:43.12 | mehale | the trunks are from good providers, they used to work ok on another asterisk installation. |
| 11:43.38 | mehale | when I use either trunk directly in a softphone, they work ok. |
| 11:44.07 | mehale | but there is this weird delay when I route the incoming call from A to B |
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| 12:45.22 | [TK]D-Fender | mehale, You're comparing a situation with no * in the middle to one without. That's an entire extra hope and processing. Go test using that same server and different codecs. |
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| 13:03.17 | mehale | [TK]D-Fender: I cant change the provider's codec. to the softphone it tanslates to whatever other codec. Between the two providers I guess it shouldnt even translate, as they both are G729. Maybe I have some wrong setting that may be causing it to translate.. |
| 13:03.35 | mehale | how can I see if agiven call is being translated? |
| 13:04.39 | [TK]D-Fender | so far I don't see you removing this translation |
| 13:04.55 | [TK]D-Fender | So you can't blame the transformation itself for the issue until you do |
| 13:05.13 | [TK]D-Fender | because otherwise it could be purely in the additional hop + server latency |
| 13:05.36 | [TK]D-Fender | And you should KNOW if it's translated or not. You define the codecs for your peers |
| 13:08.23 | Samot | mehale: What is this "delay"? |
| 13:08.33 | Samot | I mean, "delay" is vague |
| 13:08.46 | mehale | 2 seconds. what I say takes 2 seconds to reach other side. |
| 13:08.52 | mehale | quite annoying. |
| 13:08.58 | Samot | And where are you seeing this delay? |
| 13:09.04 | Samot | When you make a call or in the logs? |
| 13:09.57 | Samot | In other words, are you picking up a handset the dialing and the delay is in what you hear on the call? |
| 13:30.02 | mehale | yes, I hear the delay. SHould it be logged anywhere? |
| 13:31.36 | Samot | Have you compared this to the actual debugs and logs? |
| 13:31.51 | Samot | Hearing "delay" doesn't mean the call is being delayed in any way |
| 13:31.55 | Samot | Outside of ringback. |
| 13:32.56 | Samot | Also, phones have "timeouts" |
| 13:33.34 | Samot | For the digits being presented. If there is not "Send now" type command to tell the phone to send the digits once a pattern is matched right away, the phone will timeout waiting for new digits. |
| 13:33.42 | Samot | Usually 2-3 seconds is the standard. |
| 13:34.24 | Samot | So if you are dialing 1NXXNXXXXXX and the phone either doesn't have a pattern to match that or doesn't have a "send now" after receiving the last digit, it could be waiting for more digits before sending the call to the system |
| 13:34.28 | Samot | Thus your delay. |
| 13:35.18 | [TK]D-Fender | He's talking audio delay in conversation.... |
| 13:35.41 | [TK]D-Fender | <mehale> 2 seconds. what I say takes 2 seconds to reach other side. |
| 13:36.03 | mehale | I say Hello, the other party will only hear it 2 seconds after I said. |
| 13:36.35 | Samot | Every party? |
| 13:36.41 | Samot | Or just certain calls? |
| 13:36.54 | mehale | every party using these 2 providers. |
| 13:37.07 | mehale | As I mentioned the codec, I thought it was quite obvious that the delay was in the audio ;) |
| 13:37.31 | Samot | Delay in audio can be poor Internet |
| 13:37.46 | Samot | And not related to Asterisk at all |
| 13:37.51 | Samot | Just keep that in mind. |
| 13:39.05 | Samot | And these providers only do g729? |
| 13:40.32 | mehale | yep, they only do 729 |
| 13:40.59 | mehale | Samot, those are providers with good reputation, and my * is at hetzner.de, also has good quality network. |
| 13:41.23 | Samot | So everything from the endpoint (phone/softphone) to the provider is g729? |
| 13:41.25 | mehale | audio quality is ok, no clipping. but, it is delayed. |
| 13:41.50 | mehale | Samot: [pstn trunk g729]<-asterisk->[pstn trunk g729] |
| 13:42.50 | Samot | So wait.. |
| 13:43.06 | Samot | Its Caller --> PSTN --> Asterisk --> PSTN ---> Callee |
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| 13:43.33 | mehale | yep |
| 13:43.53 | mehale | it used to work with a much better delay, on another asterisk installation. |
| 13:44.28 | mehale | this aonther one was a PIAF, with all that crap that comes enabled and freepbx. the new one is plain asterisk with no fancy stuff. |
| 13:44.46 | Samot | Well, not just no "fancy" stuf |
| 13:45.01 | Samot | No preconfigurations that you now have to do on your own. |
| 13:45.23 | mehale | freepbx has a lot of stuf that is surplus to my needs. |
| 13:45.30 | Samot | OK |
| 13:45.32 | Samot | That's fair. |
| 13:45.39 | Samot | But FreePBX also does all the configuration of the basic stuff |
| 13:46.04 | Samot | So perhaps there is something in your configuration or setup that is different or missing due to this. |
| 13:46.55 | mehale | yes, there might be. is there anything that usually causes this audio delay? |
| 13:47.45 | Samot | Are you using chan_sip or chan_pjsip? |
| 13:49.59 | mehale | chan_sip |
| 13:50.15 | Samot | pastebin the configs for the trunks |
| 13:51.03 | mehale | one sec |
| 13:54.50 | mehale | https://pastebin.ca/4037842 |
| 13:55.49 | Samot | allow=G729 <-- this only works if there is a disallow= before it. |
| 13:56.02 | Samot | And you have no codecs defined on the other trunk |
| 13:57.12 | mehale | diallow=* and then allow=G729? |
| 13:57.34 | Samot | One of your providers is behind NAT? nat=force_rport,comedia |
| 13:57.36 | Samot | No |
| 13:57.38 | Samot | disallo=all |
| 13:57.40 | Samot | disallow=all |
| 13:57.42 | Samot | Not * |
| 13:58.14 | Samot | No real ITSP is behind NAT. |
| 13:58.19 | mehale | roger, ok |
| 13:58.41 | mehale | hm, I dont know why that nat entry |
| 13:58.53 | mehale | maybe thats from the backup conf I had. |
| 13:58.56 | Samot | Well that provider trunk is missing codec settings |
| 13:59.00 | mehale | lemme try withott that |
| 13:59.01 | Samot | And is being told it's behind NAT |
| 13:59.28 | Samot | Show your [general] section of the sip.conf |
| 14:03.37 | mehale | https://pastebin.ca/4037845 |
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| 14:04.27 | mehale | just tested now without the nat entry in the trunk, lag is now 3 secs |
| 14:04.39 | Samot | Did you set the codec? |
| 14:04.58 | Samot | Because you're allowing alaw and ulaw in the general |
| 14:05.07 | Samot | With no codec config, either of those can be done |
| 14:05.11 | Samot | So you could be transcoding |
| 14:06.29 | mehale | Samot: yes I disallowed all and allowed 729 |
| 14:06.45 | Samot | What have the providers said about this? |
| 14:06.57 | mehale | they dont help |
| 14:08.04 | Samot | 9:40:56 AM <mehale> Samot, those are providers with good reputation, and my * is at hetzner.de, also has good quality network. |
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| 14:08.15 | Samot | How can they have a good reputation when they don't help? |
| 14:08.40 | mehale | they say it is ok. |
| 14:10.37 | mehale | anyway, that delay only happens in pstn to pstn |
| 14:10.52 | mehale | not when one of the endpoints is softphone |
| 14:20.15 | mehale | what are the best practices to have a non firewalled asterisk? |
| 14:20.28 | mehale | I see that I have some ip's trying to register to mine |
| 14:20.33 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 14:22.16 | [TK]D-Fender | <mehale> what are the best practices to have a non firewalled asterisk? <- ***Firewall it*** |
| 14:22.28 | mehale | ok |
| 14:22.57 | [TK]D-Fender | You name & describe the tool used to control those things ... then ask what tool you should use.... |
| 14:23.04 | Chainsaw | You can stop a fair amount of shenigans by going TCP/TLS only. |
| 14:23.22 | mehale | when you need softphones running on mobile phones accessing it, what do you do? |
| 14:23.25 | Chainsaw | Opening UDP 5060 to the wider internet is not wise. Unless you plan to offer free telephony for all, because that's what it's going to turn into. |
| 14:23.37 | Samot | Well |
| 14:23.43 | Samot | TCP/TLS is not a real answer for that |
| 14:23.47 | Samot | At all. |
| 14:24.07 | mehale | I heard something about doorknocking, but I never found anything about that on google |
| 14:24.07 | Chainsaw | It prevents IP-spoofed scanning, which is how a lot of trouble starts. |
| 14:24.10 | Chainsaw | You'd be surprised. |
| 14:24.13 | Samot | Well |
| 14:24.26 | Samot | As someone who has been working at providers for over a decade, it's not the answer. |
| 14:24.33 | Samot | So I really wouldn't be surprised. |
| 14:25.06 | Chainsaw | It's not the only measure you take. But staying off the suckers list helps reduce your incoming scan traffic. |
| 14:25.21 | Samot | Except for the fact that there's no TLS on the PSTN |
| 14:25.47 | Chainsaw | Transposing SIP advice into PSTN just to win an argument is a bit daft. |
| 14:26.02 | Samot | So if you want to substitute more resources as the answer to a problem that has many other solutions, go for it. |
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| 14:28.47 | Samot | There are real and practical measures to take for SIP security in general before having to fall back on TCP/TLS |
| 14:29.02 | mehale | where can I read about them? |
| 14:29.09 | Samot | Google. |
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| 14:29.16 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:29.22 | Samot | Some of it is general security |
| 14:29.25 | mehale | thanks |
| 14:30.05 | mehale | what about doorknocking |
| 14:30.18 | mehale | or whatever this may be called. |
| 14:30.43 | mehale | upon registration a string would serve as a trigger to the firewall. |
| 14:30.55 | mehale | or something like that, as I understood from what I heard. |
| 14:33.21 | Samot | You need to have some sort of rules for that |
| 14:33.25 | Samot | For example... |
| 14:33.36 | Samot | I only expect traffic from ARIN (North America) IPs. |
| 14:34.00 | Samot | So all other IP blocks are denied access but I have rules and filtering... |
| 14:34.05 | mehale | ok |
| 14:34.05 | Samot | For ARIN. |
| 14:37.26 | mehale | hm, that generates a bunch of rules... |
| 14:37.34 | mehale | just for a sall country |
| 14:37.51 | Chainsaw | mehale: What you're thinking of is "port knocking", not "door knocking". |
| 14:37.59 | mehale | ok |
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| 14:45.03 | igcewieling | Todays question: in the USA from a legality point of view is using a voicemail password of 1234 considered a failure to do due diligence? |
| 14:50.58 | [TK]D-Fender | No |
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| 21:01.06 | Allears | Anyone able to provide a hint about modifying the built in unattended transfer context? (please forgive me if I don't have proper terminology) |
| 21:02.03 | file | there is no built in unattended transfer context |
| 21:02.17 | Allears | When we dial our transfer code it plays an audio file that says "transfer", we would like to stop this from happening, preferably by changing the built in method that's being used. |
| 21:02.50 | file | that isn't a context, it's the DTMF based transfer feature which has to be explicitly enabled on the call |
| 21:04.06 | Allears | @file, i see that there is a possibility to give things a different "transfer context", so I was assuming there was just a default one that it was using. |
| 21:04.27 | Allears | @file, thank you for your information, btw. |
| 21:04.40 | file | that controls what context is used for transferring to, doesn't control the DTMF feature flow (the transfer playback) |
| 21:05.10 | Allears | @file, do you know of any way to stop this "transfer" sound file from being played when we dial transfer code? |
| 21:05.31 | file | I doubt there's a way, as it's not what most people would want but dunno |
| 21:07.27 | Allears | @file, thanks again. |
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| 21:29.07 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:29.45 | [TK]D-Fender | Allears, what are you using for a phone? |
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| 21:37.55 | Dovid | how do I use media cache in asterisk? It seems to be native but when I tru to play from a url it says it cant find the file and there is no core media option (testing on asterisk 15) |
| 21:40.28 | Dovid | nm. found it |
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| 22:27.50 | Allears | D-Fender, we're using some analog phones connected through a fxs port |
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| 22:34.16 | [TK]D-Fender | Allears, on a DAHDI card? |
| 22:35.36 | Allears | D-Fender, i'm sorry I'm a newb enough not to know the answer to that question. Is there a way I can verify from within asterisk config? |
| 22:35.54 | [TK]D-Fender | You don't know what piecve of hardware your phones are pluged into? |
| 22:37.22 | Allears | D-Fender, I'm being told it is DAHDI |
| 22:37.31 | [TK]D-Fender | You can't see? |
| 22:38.12 | [TK]D-Fender | This is very odd not to have a definitive answer for if you're the one working on the box.... |
| 22:39.19 | Allears | I'm not the only one working on it. |
| 22:39.35 | Allears | and we are both very new to asterisk. |
| 22:40.35 | Allears | It is definitely a DAHDI card |
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| 22:46.08 | *** join/#asterisk chris349 (~office@104-12-70-21.lightspeed.miamfl.sbcglobal.net) |
| 22:47.04 | chris349 | Can someone tell me what is the right way to seutp a SIP trunk between 2x Asterisk servers and be able to set whatever caller ID I want? I.e.: have it authenticate by some predefined username and not by caller ID AKA SIP From: header |
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| 22:47.39 | chris349 | Because right now I have a hell of a time with it. Trunk works to make direct calls, but fails when I try to do an unattended transfer |
| 22:48.19 | [TK]D-Fender | chan_sip : set the fromuser to the username, set sendrpid & trustrpid to "yes" all on both sides |
| 22:49.33 | Allears | D-fender, @file thank you both for your help. |
| 22:50.12 | chris349 | I will try that, but is there a reason why a transfered call just fails/drops without showing some error? its very strange |
| 22:51.34 | chris349 | The moment I hit transfer the console just logs: Channel SIP/6002-00010243 left 'simple_bridge' basic-bridge |
| 22:53.20 | chris349 | And is type=peer the best to use, or would type=friend have less dropped calls? |
| 22:54.47 | [TK]D-Fender | no |
| 22:54.55 | [TK]D-Fender | auth is auth |
| 22:55.31 | [TK]D-Fender | <chris349> The moment I hit transfer the console just logs: Channel SIP/6002-00010243 left 'simple_bridge' basic-bridge <- and this isn't actual "debug" you need to have sip debug enabled and actually see the packets to know what the really process was |
| 22:55.48 | [TK]D-Fender | However You can't leve a bridge if the call wasn't accepted in the first palce |
| 22:56.02 | [TK]D-Fender | which is what would never have happened if it failed to match based on the From: |
| 22:56.07 | [TK]D-Fender | so that clearly isn't it |
| 23:25.41 | *** join/#asterisk JustSighDudes (~rashid@unaffiliated/justsighdudes) |
| 23:31.09 | JustSighDudes | Hey guys, I'm using the latest LTS Asterisk (13?). I have a question about Queues. When "Queue no answer" is set to Yes, and no one answers the call, then the call isn't logged in the CDR. When it's set to know, then the call is logged in the CDR as ANSWERED even if no one answers. |
| 23:31.22 | JustSighDudes | Any ideas on where I should look for why that's happening? |
| 23:31.38 | JustSighDudes | Even though I feel like this is expected behavior and I'm misunderstanding something |
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