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| 12:00.23 | velix | I often get such messages: "Received SSL traffic on RTP instance '0x7f80d001c4f0' without an SSL session" <-- does either the sender or receiper use SSL ? |
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| 12:47.44 | velix | Is it the other side's phone? |
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| 12:55.48 | Samot | It could be either. |
| 12:56.03 | DanFromUK | Do anyone know whether there are/were known problems with ConfBridge + PJSIP? A fairly repeatable issue is occurring where SIP audio is lost after around 30 minutes. |
| 12:56.25 | Samot | No. |
| 12:56.38 | Samot | Just one person loses audio? |
| 12:56.39 | DanFromUK | I cannot seem to find any obvious bugs on the issue tracker |
| 12:56.55 | DanFromUK | I thought one, but I have now been told about a few more. |
| 12:57.28 | DanFromUK | It could have simply been that the admin lost audio, and therefore all the non-admin thought they lost audio. |
| 12:57.37 | DanFromUK | But then it occurred to someone else later in the day |
| 12:57.47 | DanFromUK | on a separate confbridge |
| 12:58.48 | Samot | And everyone is remote to the PBX? |
| 12:59.19 | DanFromUK | Yes. And other active SIP calls (not using confbridge) haven't been affected with loss of audio. |
| 13:00.57 | Samot | Do those calls last longer than 30 minutes? |
| 13:01.08 | DanFromUK | Yes. |
| 13:01.52 | Samot | Are all the other users muted? Can they speak during this call before the audio goes out? |
| 13:02.57 | sekil | who hung up? |
| 13:03.04 | DanFromUK | In the 1st example, they had unmuted the users about 12 seconds prior to losing their own audio. So the other users realised what had happened and were able to call the admin to tell them to redial. |
| 13:03.10 | sekil | signalling should tell |
| 13:04.08 | DanFromUK | The admin did not actually lose their call, so when they were told that they werent broadcasting any more, they hung up and redialled. So technically, the admin hung up, although he hung up because his audio had stopped flowing to confbridge. |
| 13:04.51 | Samot | So only one person loses audio? |
| 13:04.54 | Samot | The rest can talk? |
| 13:05.28 | DanFromUK | It seems that way, but the issue repeated a few hours later, from a totally different admin using a totally different phone. |
| 13:05.43 | Samot | Well...you need more information. |
| 13:05.47 | DanFromUK | Actually, now that I think about it. There was an issue reported a few days ago, when someone tries to record audio into the system for playback later, they were saying that they sometimes get cut off. |
| 13:05.51 | Samot | Is the PBX on the public Internet or behind NAT? |
| 13:06.11 | sekil | this is all voip..no PSTN involved? |
| 13:06.21 | Samot | Sigh. |
| 13:06.26 | DanFromUK | Public internet, from our carrier, it won't be a NAT issue. |
| 13:06.31 | Samot | VoIP has nothing to do with the PSTN being involved. |
| 13:06.45 | Samot | DanFromUK: Untrue. There is still NAT on one side. |
| 13:06.46 | DanFromUK | PSTN -> our carrier who provides the DIDs -> internet -> asterisk |
| 13:06.47 | sekil | I meant all within pbx |
| 13:07.05 | sekil | Samot: you know what I mean |
| 13:07.25 | Samot | It doesn't matter that *I* know. |
| 13:07.32 | Samot | It matters for people who might be watching this. |
| 13:07.43 | Samot | This is a common mistake made. |
| 13:08.01 | Samot | "It's VoIP so no PSTN" is way too common. |
| 13:08.02 | DanFromUK | Samot: No NAT in my case. The SIP data is to and from public IP addresses. |
| 13:08.15 | Samot | So the phone calling the PBX isn't behind NAT? |
| 13:08.17 | Samot | At all? |
| 13:08.22 | Samot | It's out on the public Internet? |
| 13:09.27 | DanFromUK | The "phone" in this case is a PSTN user. Our carrier bridges between the PSTN<->SIP and routes SIP data from their servers to ours, all over public IP addresses with no NAT involved. |
| 13:09.39 | DanFromUK | The admin and non-admins are all dialling in via the same method. |
| 13:10.03 | Samot | So again, only one usres loses audio in the bridge? |
| 13:10.07 | Samot | Everyone else can still talk? |
| 13:10.58 | DanFromUK | It seems that way, although someone did send a text saying they couldn't hear. We are just trying to check whether the times match up with when the admin stopped broadcasting. |
| 13:11.27 | Samot | You're going to need to do more debugging |
| 13:11.44 | DanFromUK | It could simply be PSTN congestion, but I'm not 100% sure. |
| 13:12.36 | DanFromUK | Unfortunately, we don't have the disk capacity to do a full packet capture of the audio streams. I'm going to see what else our carrier can do. |
| 13:13.36 | sekil | sip capturing should be ok |
| 13:14.05 | sekil | check session timers etc. |
| 13:14.16 | Samot | Or you know. Make test calls to replicate the issue. |
| 13:14.34 | Samot | Because if this is a "common" problem that is happening based on user reports. You should be able to replicate it. |
| 13:17.39 | DanFromUK | yeh. I'll set up a test PSTN call going. |
| 13:20.02 | sekil | is there a min-se and/or session expires header on invites from provider |
| 13:22.12 | DanFromUK | Been with them for 10 years. First time I've seen this issue. But it's also the first time I've been heavily using ConfBridge. |
| 13:22.32 | Samot | What are the resources on the server? |
| 13:22.49 | DanFromUK | server specs? |
| 13:22.56 | Samot | Yes |
| 13:25.58 | DanFromUK | The asterisk machines are running under KVM with 4 virtual CPUs. |
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| 13:26.08 | DanFromUK | <PROTECTED> |
| 13:26.15 | DanFromUK | During these ConfBridge calls, they were nowhere near their previous peak. |
| 13:28.29 | DanFromUK | If the issue previously mentioned (of potential lost calls when performing an audio recording, outside of ConfBridge), the issue is affecting multiple Asterisk instances. And therefore more likely to be a carrier issue, or an issue with our firewall, network etc. |
| 13:28.56 | DanFromUK | Hence why I was checking whether anyone could recall an issue with lost audio specifically with ConfBridge. |
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| 15:01.09 | velix | DanFromUK: session-timers=refuse |
| 15:01.17 | velix | It worked for me on chan_sip |
| 15:01.42 | velix | Some calls ended after 15 minutes |
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| 15:28.12 | _ganapathi_ | am using confbridge. first creating confbridge with agent channel and dialling customer and connecting into confbridge. but before dialling customer want to execute play customer name to the user. but if i play using redirect and connecting again to confbridge then lost admin access of confbridge. |
| 15:28.14 | Samot | velix: Dan's using PJSIP. |
| 15:31.53 | _ganapathi_ | any way to send DTMF by script to the channel ? |
| 15:32.36 | [TK]D-Fender | AMI clearly... |
| 15:40.22 | velix | Samot: Is there no substitude for this? |
| 15:40.51 | Samot | Yes there is but that setting doesn't exist in PJSIP. |
| 15:41.28 | Samot | Timers can be enabled (Default) or not. You can set the min and max session timer. |
| 15:41.53 | velix | Ah okay. |
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| 16:33.18 | _ganapathi_ | local channel & SIP channel is created for user to be on confbridge. if am sending DTMF to SIP channel then confbridge dtmf action not working.. |
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| 17:07.29 | velix | Should I always add a "exten => 3001,n,Hangup()" hangup? |
| 17:12.27 | DanFromUK | velix: why wouldn't you? |
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| 17:13.10 | Helenah | Hi |
| 17:13.11 | velix | DanFromUK: On the end of an extension, I never had problems with leaving it out (for mistake) |
| 17:14.23 | DanFromUK | velix: I am not an expert. I think it's best to including it, just in case you have more dialplan somewhere which could accidentally continue processing. |
| 17:14.36 | velix | DanFromUK: I agree. Thanks |
| 17:14.45 | Helenah | Okay so... I have a guest WiFi SSID and any guest connecting to this SSID I want to give access to an Asterisk box. The SSID has its own NAT. I've forwarded the SIP and RTP ports so devices from behind the NAT can access the Asterisk box which they can. The issue I'm having now is the guests can't transmit or receive voice. |
| 17:15.08 | Helenah | I'm aware Asterisk is a pain when there is a NAT involved however this is a requirement. |
| 17:15.35 | DanFromUK | Helenah: Get a public IP address for the Guest SSID? |
| 17:16.31 | Helenah | The Asterisk box is on a LAN, all devices on that LAN can access the Asterisk box fine. However there is a guest SSID which is NATed so that guests can't just wonder around the LAN but only have access to what I forward. |
| 17:16.51 | Helenah | So no public IP needed. |
| 17:17.37 | Helenah | Basically guests are on their own isolated network. |
| 17:17.59 | DanFromUK | Can you multihome the asterisk box so it gets an IP on the main LAN and an IP on the guest LAN? |
| 17:18.30 | igcewieling | I don't see the need for NAT unless the WiFi is at a different location than the Asterisk box. |
| 17:18.49 | DanFromUK | I think to do that, you need to create two PJSIP transport configurations. |
| 17:19.09 | DanFromUK | one for each LAN (main and guest) |
| 17:19.27 | igcewieling | Asterisk will reply from the same address as the packet was received from. |
| 17:20.46 | DanFromUK | igcewieling: Not strictly true. From experience, the asterisk server binds to 0.0.0.0, and the NIC has two IP addresses on the same subnet, the outbound asterisk traffic comes from the default IP for that NIC. |
| 17:21.08 | Helenah | Right I'm confused now. |
| 17:21.11 | DanFromUK | if you set a specific IP, instead of 0.0.0.0, in the asterisk config, then it locks it down. |
| 17:21.23 | DanFromUK | Helenah: Give the asterisk server an IP on the guest network. |
| 17:21.30 | igcewieling | that kind of network setup is a dumb idea and might indeed require multiple transports. |
| 17:21.51 | igcewieling | Use two different networks without NAT involved. |
| 17:22.07 | Helenah | How would I do that? And why do I need to do that? I can connect to the box via the guest network so why can't I just transport audio? |
| 17:22.09 | igcewieling | guest LAN, and private LAN |
| 17:22.48 | igcewieling | Helenah: no way to tell without spending an hour understanding your setup. |
| 17:23.46 | Helenah | I have a WiFi AP, it has an SSID on it which is private and uses the LAN's subnet and it has a guest SSID which uses its own subnet and a NAT. |
| 17:28.20 | igcewieling | what are those subnets? |
| 17:29.08 | *** join/#asterisk Helenah (d9a30ba3@unaffiliated/iveeee) |
| 17:29.16 | Helenah | Damnit, did anyone respond? |
| 17:29.59 | igcewieling | you are going to have to start providing specific info like subnet info to do anything more than speculate. |
| 17:32.37 | Helenah | Private LAN (10.1.1.1) Guest LAN (10.0.1.1) |
| 17:33.45 | Helenah | My Asterisk box is 10.1.1.254 |
| 17:52.53 | igcewieling | does your asterisk box have two network interfaces or do you have a vlan capable switch? |
| 18:35.33 | MLC | One of our employees that is newly working at home has a problem. See https://pastebin.com/H4vFw4Mj I use that call file to initiate a call to her land line and then the dial plan dials an outside line and bridges them together. The call connects and there is 2 way audio, but she doesn't hear any ringing between the Playback() and when the remote end answers the call. Any ideas? |
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| 19:08.55 | MLC | The plot thickens: bridge_native_rtp.c: Remotely bridged 'PJSIP/voip.ms-tollfree-00012221' and 'PJSIP/voip.ms-tollfree-00012223' - media will flow directly between them |
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| 19:09.24 | MLC | That explains why the call drops after 1 minute, because my asterisk times the call out after 60 seconds of no RTP |
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| 20:33.29 | MLC | Just in case anyone is watching, I "solved" the "not ringing" problem by change R to r(ring) in the dial plan. It doesn't really mean that the remote end is ringing, but it makes the agent happy' |
| 20:37.55 | igcewieling | have you tried adding ,noanswer to your playback instead? |
| 20:38.58 | MLC | I haven't, but I will. I didn't even not that option existed! |
| 20:39.07 | MLC | *know |
| 20:39.12 | igcewieling | You could instead try running Ringing(). |
| 20:39.34 | igcewieling | core show application playback |
| 20:39.39 | igcewieling | ALWAYS read the app docs. |
| 20:39.50 | MLC | hangs his head in shame |
| 20:41.17 | igcewieling | Ideally the call is not answered until the callee answers. That isn't always practical, but worth trying. |
| 20:41.46 | MLC | good point |
| 20:47.23 | MLC | I fixed the no RTP by setting rtp_timeout = 0 in my pjsip.conf. That seems like kind of a bug if the call is remotely bridged? |
| 20:48.10 | file | https://issues.asterisk.org/jira/browse/ASTERISK-28774 |
| 20:48.27 | MLC | sweet. I need to upgrade. |
| 20:48.31 | file | it's not in a release yet |
| 20:48.35 | file | it was fixed 6 days ago. |
| 20:48.48 | MLC | I'm always a day late and $$$ short |
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| 22:17.03 | velix | I'm trying to use Goto to redirect an incoming call to a conference. But the call is directly hung up. Can anyone help me here? https://bpaste.net/raw/3CNA |
| 22:24.44 | velix | ah, got it |
| 22:32.35 | _ganapathi_ | am using confbridge on asterisk 17. here i want to perform some dialplan execution to admin user on confbridge in background without leaving channel without losing admin permission. same is configured with dtmf input on confbridge.conf but only works when admin user pressed DTMF. but we want to perform action automatically. any possibility ? |
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| 22:35.04 | Samot | What do you mean automatically? |
| 22:35.12 | Samot | They are in the confbridge. |
| 22:35.14 | _ganapathi_ | through AMI |
| 22:35.21 | Samot | OK but how is that automatic? |
| 22:35.48 | _ganapathi_ | according to some event |
| 22:35.54 | _ganapathi_ | need to play |
| 22:36.03 | Samot | They are in the confbridge, what event would you be looking for? |
| 22:36.56 | _ganapathi_ | admin user on confbridge. then i will dial the customer continously and patching into confbridge. here want to play customer name before dial/while dialing. |
| 22:37.12 | *** part/#asterisk MLC (~MLC@63.249.40.11) |
| 22:37.31 | Samot | Doubt that can happen. |
| 22:37.45 | Samot | Outside of updating the MoH in the bridge but I don't think you can do that either. |
| 22:38.27 | _ganapathi_ | without playing customer name , i can able to do that. but to play to the channel then it's leaving confbridge, then again i need to put it back. |
| 22:39.10 | _ganapathi_ | if am putting back also then i lost admin access of confbridge in-case if any other user joined on the confbridge |
| 22:40.01 | Samot | That's what I was getting it. |
| 22:40.12 | Samot | While they are in the bridge, you can't do what you are looking for. |
| 22:40.44 | Samot | So why don't you do something like Park the admin, when the call connects push them both to the bridge. |
| 22:40.53 | _ganapathi_ | oooh Ok. |
| 22:41.08 | _ganapathi_ | park the admin into ? |
| 22:41.17 | Samot | A parking lot. |
| 22:41.31 | Samot | If you need to hold the admin while you dial the end user... |
| 22:42.00 | _ganapathi_ | how do i do that ?. i don't have idea ! |
| 22:42.12 | _ganapathi_ | which action. any example? |
| 22:42.16 | Samot | Look at the Park features and functions. |
| 22:42.23 | Samot | You need to setup the lot |
| 22:42.39 | Samot | Then you can use dialplan functions or AMI actions to do what you need. |
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| 22:43.40 | MLC | Spent the last hour chasing down why a certain call wouldn't go through. There was a garbage character in extensions.conf that didn't show up in my editor. Good grief! |
| 22:43.53 | _ganapathi_ | ok i will try that. will u pls give exact scenario on the same case. |
| 22:45.10 | Samot | Admin dials into Asterisk, you park them in the lot, dial the second party when second party answers, pull admin from lot, play announcements, push to confbridge. |
| 22:45.37 | Samot | Use the confbridge functions to set needed settings on each channel before putting them in the confbridge. |
| 22:49.28 | _ganapathi_ | thanks. |
| 22:49.49 | _ganapathi_ | admin need to go first into confbridge rgt |
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