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| 08:40.10 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); 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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| 13:50.39 | *** join/#asterisk lagzilla (lagzilla@2600:3c03::f03c:91ff:fe6e:5ed9) |
| 13:55.24 | lagzilla | I don't know all the terminology but is it possible to do something like exten => s,1,Page(SIP/${SIPACCOUNT}@{$SERVER}@members) |
| 13:56.47 | file | to accomplish what? |
| 13:59.49 | [TK]D-Fender | You pass Tech just like you would in a DIal |
| 14:01.08 | Samot | Except that would be an improper Dial() string. |
| 14:01.32 | Samot | SIP/${EXTEN}@${SERVER}@members is invalid. |
| 14:01.34 | lagzilla | We have a HA setup and currently the dial string is Local/',username,'@page-members2, So members on the 2nd server don't get called. So I'm trying to figure out how to have it as SIP/account@server |
| 14:02.03 | Samot | If you have HA why would the members on the second server get called? |
| 14:02.06 | Samot | That's not how HA works. |
| 14:02.30 | lagzilla | soory meant load balanced |
| 14:02.38 | lagzilla | and we're not using an SBC to manage it |
| 14:02.46 | Samot | That's even worse. |
| 14:03.11 | lagzilla | lol yeah but that's how it is |
| 14:03.57 | Samot | I'm pretty sure it doesn't have to be. |
| 14:04.05 | Samot | It's an inadequate solution. |
| 14:05.04 | lagzilla | Maybe but I'm a junior employee I can't change anything I just need to make it work lol |
| 14:05.19 | lagzilla | Though I have no idea what I'm doing to begin with lol |
| 14:05.44 | Samot | Well that Page() string isn't going to work. |
| 14:06.19 | lagzilla | yeah I'm trying to read around but I'm too dumb to figure it out lol |
| 14:06.35 | lagzilla | what's the '@page-members2 part of the dial string called? |
| 14:06.53 | Samot | It's the context. |
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| 14:07.47 | Samot | Local/100@page-members2 will go to [page-members2] context looking for a match for 100 |
| 14:08.00 | Samot | To execute the dialplan. |
| 14:08.14 | lagzilla | Can I do that with SIP/account@IP |
| 14:08.27 | Samot | That will dial out of the system |
| 14:08.31 | Samot | To account@ip |
| 14:09.26 | lagzilla | Can I just do page(SIP/account@ip) then? |
| 14:09.37 | Samot | Where is it going to go? |
| 14:10.08 | lagzilla | ext 100,101,102 on server a, and 200,201,203 on server b |
| 14:10.35 | Samot | So you have the routing setup on server b to accept these calls? |
| 14:10.44 | Samot | And then actually PAGE the devices at Server B? |
| 14:10.55 | Samot | You can't do this just across servers like that |
| 14:11.36 | Samot | This is why this is a poor setup solution for this. |
| 14:11.46 | Samot | You really can't cross system somethings. |
| 14:11.52 | Samot | Paging is going to be one of them. |
| 14:13.06 | [TK]D-Fender | He can't hit the peers direct, but if his dialplan is set for it the calls can go through right. |
| 14:13.07 | lagzilla | Yeah it works for extensions/queues |
| 14:13.14 | [TK]D-Fender | But the receiving end has to do everything extra |
| 14:13.22 | [TK]D-Fender | CAN hit* |
| 14:13.25 | [TK]D-Fender | oops |
| 14:13.33 | [TK]D-Fender | I double misread that |
| 14:13.38 | [TK]D-Fender | yeah, first was right |
| 14:16.20 | Samot | Except it's a page. |
| 14:16.37 | Samot | So now it has the Channel on PBX B and then another Channel for the device off PBX B. |
| 14:16.42 | Samot | What's answering that page? |
| 14:16.49 | Samot | How is that going to get through? |
| 14:16.55 | Samot | Are there Auto Answer headers? |
| 14:16.59 | [TK]D-Fender | That'ws up to the dialplan on the server clearly |
| 14:17.06 | Samot | See there is a lot to go through for this. |
| 14:17.07 | [TK]D-Fender | but nothing has to implicitly get in the way |
| 14:17.16 | Samot | What about the Auto Answer headers?! |
| 14:17.31 | Samot | How do those go from PBX A to PBX B to the device? |
| 14:17.34 | [TK]D-Fender | as I said if he wan't that its on B's dialplan for handling it |
| 14:17.49 | Samot | How about load balancing a PBX between two system is a bad idea. |
| 14:17.59 | Samot | Where User A can end up on either PBX |
| 14:18.06 | Samot | while User B can also end up on either PBX |
| 14:18.16 | [TK]D-Fender | That's another matter your experience can cover.. |
| 14:18.23 | [TK]D-Fender | the basic page is possible if he controls it all |
| 14:18.32 | Samot | I know. |
| 14:18.39 | Samot | It's all the extra BS work to make it happen. |
| 14:18.41 | [TK]D-Fender | His other complications you've chimed in on already... |
| 14:18.42 | Samot | Just like anything else. |
| 14:19.12 | lagzilla | It looks like I can jam in exten => *,1,SIPAddHeader(Call-Info: <sip:10.1.1.171>\;answer-after=0) |
| 14:19.31 | Samot | lagzilla: Where are you jamming that in? |
| 14:19.57 | lagzilla | in the [page] context in my dialplan |
| 14:20.04 | lagzilla | Going off of https://www.voip-info.org/asterisk-howto-dial-plan/ |
| 14:20.05 | Samot | On PBX A? |
| 14:20.10 | lagzilla | yeah |
| 14:20.12 | Samot | OK |
| 14:20.18 | Samot | So how will PBX B know about it? |
| 14:20.20 | [TK]D-Fender | that has to be on B for those calls |
| 14:20.27 | Samot | That was my original question. |
| 14:20.28 | lagzilla | oh |
| 14:20.41 | Samot | You have to do all this again on PBX B |
| 14:20.42 | [TK]D-Fender | it doesn'tmagically carry on when B gets the call and dials out again |
| 14:21.02 | lagzilla | Oh I would of thought the SIP header just get's passed along |
| 14:21.09 | lagzilla | guess I'm wrong |
| 14:21.10 | Samot | lagzilla: All your users should be on the same system. |
| 14:21.20 | Samot | They should not be split between the two |
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| 14:22.31 | lagzilla | I'm not management though I can't change what's in place regardless of being a bad setup, from everything I've read we should be handling this tough a load balancer but it seems that wasn't the solution implemented. |
| 14:22.43 | Samot | Even then that's still not the answer. |
| 14:22.48 | Samot | What are you load balancing? |
| 14:22.55 | Samot | How many users are on this system? |
| 14:23.09 | lagzilla | 300ish |
| 14:23.16 | Samot | For a single company? |
| 14:23.22 | lagzilla | yeah |
| 14:23.37 | Samot | Well this is the wrong solution. |
| 14:23.46 | Samot | And it's going to make supporting it more of a job than it needs to be. |
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| 14:24.00 | Samot | It's going to raise more support problems than it's solving. |
| 14:24.09 | lagzilla | lol justifies my paycheck I guess |
| 14:24.24 | Samot | Problems that are beyond your scope. |
| 14:25.04 | lagzilla | And my knowledge base *cries* |
| 14:27.17 | Samot | So the bosses want you to solve this issue but at the same time you're saying you can't change what is being done... |
| 14:27.30 | Samot | What is solving the issue requires changes to what is currently in place? |
| 14:27.39 | Samot | What then? |
| 14:27.40 | lagzilla | They want it solved while maintaining the current deployment |
| 14:27.53 | Samot | Then they need to step back and look at it overall. |
| 14:28.04 | Samot | Because the communications between the two systems needs to be addressed. |
| 14:28.23 | Samot | Things like Paging, Parking, Conferences, Queues... |
| 14:28.31 | Samot | Those are things that really can't be a "shared systems" like this. |
| 14:29.28 | lagzilla | Most of those work somehow, it's just paging and remote pickup that are broken |
| 14:29.43 | Samot | What remote pickup? |
| 14:29.59 | Samot | You mean someone on PBX B getting the page from PBX A? |
| 14:30.05 | lagzilla | We have like **exten and you can pickup a ringing ext |
| 14:30.16 | Samot | Call Pickup isn't going to work either. |
| 14:30.21 | Samot | Because you have calls on TWO SYSTEMS. |
| 14:30.36 | Samot | You can't Call Pickup a call from PBX A on PBX B. |
| 14:30.40 | Samot | Just not going to happen. |
| 14:31.36 | Samot | PBX A and PBX B have no clue on each others calls. |
| 14:31.41 | Samot | Channels or anything. |
| 14:31.55 | Samot | Everything is based on the channels in the PBX. |
| 14:32.25 | Samot | This deployment setup is causing these issues. |
| 14:32.37 | Samot | These are all issues that exist due to this setup. |
| 14:32.38 | lagzilla | That's all in the DB which is usually how we manage channels in exten.conf |
| 14:32.46 | Samot | That doesn't matter. |
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| 14:33.00 | Samot | PBX B would need to know the channel on PBX A to pick it up. |
| 14:33.16 | Samot | And even then, that still might not work. |
| 14:33.56 | Samot | Because user on PBX B will do a Pickup, that will dial PBX B then it has to dial PBX A to get to it and then grab that channel. |
| 14:34.01 | Samot | It's a complete mess. |
| 14:35.56 | Samot | You cannot treat load balancing in SIP the same way your do with HTTP/S or other TCP based services. |
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| 16:12.03 | craigify | I designed a multi tenant PBX system in Asterisk and NodeJS for a company where I am the CTO. We have different classes of Asterisk servers. The customer facing SIP servers that interact with SIP in the field do minimal internal call logic. For every call placed to a customer SIP server, it immediately sends that call to an internal cluster of PBX systems, which then determines what to do. If it sends it to the PSTN, it goes to another |
| 16:12.03 | craigify | cluster of PSTN connected Asterisk. If it goes to another SIP device, it determines what customer SIP server that device is registered to first, then goes back to the SIP server, then to the SIP device |
| 16:13.08 | craigify | no out of the box PBX would do it, and just taking one and trying to load balance it probably would be equal amount of work, I'd wager a guess. |
| 16:13.18 | craigify | Just some insight on what it takes to do something like that, is all |
| 16:13.48 | Samot | Well that is a completely different scenario. |
| 16:14.01 | Samot | This is a company with 300 users and they're load balancing between two PBX systems. |
| 16:14.51 | Samot | In your scenario, using Asterisk as switches/gateways like that isn't the best option either. |
| 16:15.01 | Samot | That's where actual switches/SBCs would be used. |
| 16:31.30 | avb | hey guys. in case if you receive inbound call, and then you want to forward call to another phone number, how would you create 2 separate cdrs for those calls? |
| 16:32.02 | avb | i have tried forkcdrs but for some reasons im getting 2 cdrs with the same record data |
| 16:39.50 | Samot | There will already be separate CDRs. |
| 16:40.10 | Samot | Inbound Call --> Asterisk <<< Generates CDRs for what that calls does. |
| 16:40.30 | Samot | Asterisk --> Fwd Number <<< Generates CDRs for that that call does. |
| 16:45.05 | avb | (facepalm) |
| 16:45.08 | avb | im working too much |
| 16:45.20 | avb | you are correct |
| 16:45.31 | Samot | Well it's still up to you to mark that call properly. |
| 16:45.41 | Samot | Asterisk doesn't really have a concept of "call forwarding" |
| 16:45.54 | Samot | Your inbound call is going to hit the PBX. |
| 16:46.17 | Samot | You are going to decide then if you're calling the endpoint or "call forwarding" the call. |
| 16:46.51 | Samot | If you call the endpoint first, then that Dial() is the call. |
| 16:47.14 | Samot | If it's CFU, then a NO ANSWER is just going to tell Asterisk to Dial() again. |
| 16:47.29 | Samot | Based on your dialplan. So it's two separate calls overall. |
| 16:48.04 | Samot | So now you'll need something to link the CDRs for the inbound call, then call to the endpoint and then then call to the CF destination. |
| 16:48.23 | Samot | then=the |
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| 16:54.22 | avb | still beautiful |
| 16:54.27 | avb | uniqueid is the same |
| 16:54.32 | avb | dst is the same |
| 16:54.56 | avb | if I will call Set(CDR(xyz)=123) it will update both records |
| 16:56.57 | Samot | Yeah, I just use the initial From Tag for that. |
| 16:57.02 | Samot | Same concept. |
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| 18:10.57 | avb | this is not even fun :) |
| 18:11.13 | avb | i wonder how that happend that i never saw this problem before |
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| 20:56.47 | jmordica | Hi all. Asterisk 16. What are some ways to optimize queues for more performance on |
| 20:56.50 | jmordica | high volume systems? Besides simultaneous ringing. |
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