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| 00:16.26 | *** 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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| 09:16.04 | henkebe | Hi! |
| 09:16.04 | henkebe | I am using Asterisk 1.4 with queues and agents in mysql (dynamic realtime). There is a web application connected to this setup that uses rawman/AJAM to send manager commands. It was developed a long time ago, but have worked well. But now I have stumbled upon a problem with agent status (paused/unpaused). If I for example change an agents status to unpaused in the database, this is not |
| 09:16.04 | henkebe | reflected in Asterisk until a call comes in. A simple example: we have one queue, one agent who is paused both in asterisk and in mysql. A call comes in and is placed in the queue. We change the agent status in the database to unpaused. Nothing happens and the incoming call is still waiting in the queue. If another call comes in, this trigger an update from the database and the first call |
| 09:16.04 | henkebe | is answered by the now unpaused agent. If i do "queue show <queuename>" in asterisk cli, this triggers an update too. But I am looking for some way to force asterisk to reload the agent status. I have tried all manager commands I can think of that could do this, but nothing works. Does someone have an idea of how I can force Asterisk to read from the queuemember-table and update the statuses |
| 09:16.04 | henkebe | of the agents? |
| 09:16.37 | Reinhilde | Too old. |
| 09:18.55 | henkebe | Yes, but unfortunately I cannot upgrade for the moment, I will have to get it working somehow. |
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| 09:51.49 | mvanbaak | 1.4... nice |
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| 16:30.02 | polyfonix | Hello people |
| 16:31.57 | polyfonix | definitely a noob question, but was wondering whether it is possible to modify the channel name from extensions.conf ? Also how does asterisk determine the channel name ? |
| 16:33.11 | file | you can't, and it's generated based on however the channel driver wishes to create it |
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| 16:38.19 | polyfonix | Ok, Is there someway i can dig deeper into how the channel driver is creating it ? Currently my channel driver is SIP . |
| 16:38.20 | [TK]D-Fender | it's based ont he tech, plus the kind of suffix it uses |
| 16:39.11 | file | it's in the source code |
| 16:40.02 | file | chan_sip.c is so large I can't link it on Github, but ast_channel_alloc is the function that is called |
| 16:40.51 | igcewieling | polyfonix: WHY do you want to change it? |
| 16:41.32 | polyfonix | Thanks file |
| 16:43.48 | polyfonix | igcewieling Currently I'm trying to create a build a sip - multitenant platform. While doing this I would have clients with sip:user1@domainA, sip:user2@domainB. Internally I've a sip provider who would handle all these calls, consider <provider> |
| 16:44.28 | Samot | And how does changing the channel name help with this? |
| 16:46.31 | polyfonix | So if i need to connect both of these clients I need them to connect through <provider> . Currently the channel name is being set to domainA or domainB . I'd rather prefer the channel variable to be set as <provider>, rather than domainA or domainB . We also have some legacy code that picks up different stuff based upon the channel variable. So if |
| 16:46.32 | polyfonix | the channel variable is set to the provider running these calls It would really help in reducing the amount of legacy code change. |
| 16:46.32 | igcewieling | polyfonix: It is rarely good to change Asterisk to suit your needs, it is better to find a way without modifying things like channel names. |
| 16:46.56 | Samot | What does the provider need to connect these calls? |
| 16:47.19 | igcewieling | polyfonix: you can set variables on a per-peer basis. see setvar= in sip.conf.sample. |
| 16:47.37 | igcewieling | it would be better to use pjsip if you are starting from scratch |
| 16:47.41 | Samot | s/Why/What/ |
| 16:49.52 | Samot | polyfonix: I believe you might be approaching this wrong based on what you've said so far. |
| 16:50.53 | polyfonix | igcewieling we aren't starting from scratch. I'm a new dev in this project without much idea about asterisk and the PBX stack, so I'm trying to figure out things. Also since we would have multiple domain names coming into picture it wouldn't be scalable to have per-peer basis variable setup. This would mean everytime we onboard a new client we woul |
| 16:50.53 | polyfonix | d have to setup the sip.conf variables, wouldn't it ? Rather I would like this provider to handle all the calls for the domains. I'm not sure if I'm making sense. |
| 16:51.05 | igcewieling | polyfonix: on the multi-tennant platform I use, a 3-digit "tennant" code is added to the extension, so extension 1234 on tennant 987 would need to register as user 9871234. |
| 16:51.45 | Samot | polyfonix: this is is a true multi-tenant system then yes, each tenant would have its own configs. |
| 16:53.41 | Samot | polyfonix: Who handles the PBX level feature stuff? Like voicemail, call forwarding, etc? You or the provider? |
| 16:54.25 | polyfonix | Samot We handle the PBX level features stuff. The provider handles the media, client sdks . |
| 16:54.46 | Samot | You would be handling the media if it's on Asterisk. |
| 16:55.05 | Samot | And then you would need to isolate your tenants in Asterisk. Otherwise you end up with cross dialing |
| 16:55.13 | Samot | One user in another tenant's voicemail, etc. |
| 16:55.57 | Samot | I'm also still trying to figure out what Tenant A and Tenant B need the PSTN to call each other.... |
| 16:56.05 | Samot | Damn it. |
| 16:56.12 | Samot | s/What/Why/ |
| 16:57.26 | polyfonix | Well this entire thing is confusing enough for me, and it's difficult explaining it to somebody else. |
| 16:57.55 | Samot | So is Asterisk the only thing being used? |
| 17:00.41 | polyfonix | So TenantA@domain1.com and TenantB@domain2.com are connected to provider. TenantA calls provider saying he wants to connect to <myPhoneNumber> . <provider> then dials to my <myPhoneNumber> which resides on <myAsterisk> . Now my PBX software knows that when tenantA dials to <myPhoneNumber> I should connect him to TenantB@domain2.com. I then dial to |
| 17:00.41 | polyfonix | <PROTECTED> |
| 17:01.04 | Samot | That's a waste. |
| 17:01.35 | polyfonix | Yeah I know, because of this we have latency related issues. |
| 17:01.45 | Samot | And cost waste issues. |
| 17:02.14 | Samot | You should not be sending calls to the PSTN that are essentially on-network. |
| 17:02.44 | polyfonix | true. But seemingly the management has taken a decision to go this way due t not having SDK related experience. |
| 17:02.56 | Samot | I'm not sure what SDK has to do with this. |
| 17:03.05 | polyfonix | These are SIP calls. |
| 17:03.10 | polyfonix | they aren't PSTN calls. |
| 17:03.10 | Samot | I know what they are. |
| 17:03.24 | Samot | PSTN = Public Switched Telephone Network |
| 17:03.35 | Samot | IE. How calls route to each other. |
| 17:03.56 | Samot | It doesn't matter if it's copper or SIP a call from you to ATT or another provider goes over the PSTN. |
| 17:04.21 | Samot | Sending the call to your provider means they are most likely charging you for the termination and origination of the call. |
| 17:04.27 | polyfonix | Yeah I'm from India, here we refer to PSTN as the landline/mobile phones. Here in India, we cannot intermix, SIP and PSTN . |
| 17:04.45 | Samot | And where is this project out of? |
| 17:04.56 | polyfonix | India |
| 17:06.38 | Samot | So changing the channel names, bad idea. |
| 17:06.47 | polyfonix | By SIP call I mean it's like a skype audio call. |
| 17:06.56 | Samot | I know what SIP calls are. |
| 17:07.24 | polyfonix | I meant no offence, but that is why we need the SDK's . |
| 17:07.37 | igcewieling | polyfonix: what exactly do the SDKs do? |
| 17:07.58 | Samot | There is a difference between the softphone client and the voice network. |
| 17:08.00 | polyfonix | The SDK's are like clients to the provider. They can accept and make calls. |
| 17:08.08 | polyfonix | yeah just like a softphone client. |
| 17:08.14 | Samot | Which connect to Asterisk |
| 17:08.20 | igcewieling | why not use any one of dozens of softphones which already exist |
| 17:08.26 | polyfonix | In this case the SDK is a softphone client which connects to providers SDK. |
| 17:08.35 | Samot | So why is Asterisk involved? |
| 17:08.48 | igcewieling | It isn't as far as I can tell 8-| |
| 17:08.53 | Samot | Right. |
| 17:10.16 | Samot | I'm not sure how the sofphone client connects to the providers SDK but Asterisk is doing all the PBX features. |
| 17:10.33 | Samot | The softphone client would need to be connected/talking to Asterisk not the provider. |
| 17:10.39 | polyfonix | Well, billing and other post processing activities after the call take place through systems connected to our PBX. So that is why we would want to route calls through our asterisk software so that other activities can also happen unhindered. |
| 17:11.07 | Samot | And there is an Asterisk expert on the project? |
| 17:11.24 | igcewieling | Sounds like the ingredients for a disaster. |
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| 17:13.06 | polyfonix | God no, there is no asterisk expert in this project. And well the other dev's are also apprehensive about this. |
| 17:13.17 | Samot | Well then you should get one. |
| 17:13.18 | polyfonix | Mgmt want to Go to market asap . |
| 17:13.20 | Samot | Because you need one. |
| 17:13.26 | Samot | Not going to happen. |
| 17:13.35 | Samot | I did a project like this a few years back, sounds the same. |
| 17:13.56 | Samot | Everything was so focused on the app/SDK that not of the actual telephony/PBX stuff worked properly |
| 17:13.59 | polyfonix | I know where you are coming from, I raised this to my manager in the last one on one. and he was of the opinion that I should try to become that person :/ |
| 17:14.10 | Samot | And when they brought me in, I said things needed an overall and they refused. |
| 17:14.20 | Samot | Because it meant re-writing a bunch of the application. |
| 17:14.46 | Samot | Due to the fact the app was designed with no voice/telephony experience in it or an idea of what the platform could do. |
| 17:15.13 | Samot | Management can go to market as fast as they want as long as they want to crash and burn. |
| 17:15.13 | polyfonix | Thanks @samo |
| 17:15.26 | Samot | That project failed, BTW. |
| 17:15.34 | Samot | $600,000 USD later. |
| 17:15.36 | polyfonix | Thanks Samot and @igce |
| 17:17.01 | polyfonix | Thanks Samot and igcewieling, I really must head home now as it's getting late and cold. catch you guys later. |
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| 17:23.29 | Samot | igcewieling: I'm with your assessment. |
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| 19:29.42 | [sr] | howdy, there's a command that show's me the headers or some headers of the sip peer, i want to know the trademark and model of the device of that sip peer |
| 19:29.48 | [sr] | sip show peer 111 |
| 19:29.53 | [sr] | dont quite remember |
| 19:30.45 | [sr] | sip show user 111 doesnt provide that info |
| 19:31.38 | [sr] | sip show peer 111 |
| 19:31.40 | [sr] | thanks :P |
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| 19:43.43 | lagzilla | grabbing one of the option packets is likely the easiest |
| 19:43.45 | lagzilla | way |
| 19:47.44 | lagzilla | https://www.asanka.me/2015/11/asterisk-phone-inventory-useragent-list/ |
| 19:47.49 | lagzilla | oh maybe you can get it |
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| 20:21.48 | Samot | [sr] sip show peer 111 will show the UA of a registered contact. |
| 20:21.51 | Samot | Useragent : Cisco/SPA112-1.3.5(004) |
| 20:22.18 | [sr] | Samot: that, thanks, i got it from there |
| 20:55.18 | igcewieling | too weird for an error message? "no faxes found. perhaps they have gained sentience and are plotting revenge." |
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| 21:26.34 | Samot | Or "No Faxes Found. Perhaps they're trying to discover their true self." |
| 21:27.01 | Samot | You know, since they're facsimile's |
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