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00:19.31 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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08:55.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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09:29.36 | td34 | driving me insane!! :P |
09:47.44 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
09:48.29 | samwierema | What mailing list is best to track security issues in Asterisk? asterisk-dev? |
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10:11.06 | file | you mean to be notified of them? |
10:13.19 | file | we announce on -users, -dev, -announce, Twitter, and the topic here is updated with the security releases as the latest |
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10:59.04 | samwierema | file: thanks, I missed the "See all lists" link on the website and couldn't find the -announce list |
11:12.58 | *** join/#asterisk dnit (673399be@gateway/web/freenode/ip.103.51.153.190) |
11:13.15 | dnit | Hi |
11:13.33 | dnit | How can I analyze early media in a call |
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12:56.14 | Samot | Analyze? |
12:56.16 | Samot | What? |
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14:05.13 | igcewieling | dnit: wireshark should work. |
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14:17.14 | dnit | igcewieling: I mean in asterisk dialplan. Like AMD application does some processing after call starts. I found this app https://github.com/phatjmo/app_cpa but cant figure out how to use it on early media |
14:18.46 | dnit | samot: sorry I mean process not analyze |
14:18.46 | igcewieling | bad questions lead to bad answers. |
14:19.57 | igcewieling | I'm not aware of anything in Asterisk to do that. Good luck. |
14:20.35 | dnit | :( |
14:20.43 | igcewieling | My former employer charged thousands of dollars per month to customer needing custom AMD. |
14:21.35 | igcewieling | What are you trying to accomplish with this crazy idea? |
14:22.55 | dnit | We are observing a lot of calls which are transmitting " this number is out of service" in early media instead of proper sip signalling. |
14:23.25 | dnit | Our system currently mark these calls as No Answer calls. This is leading to unhappy customers. |
14:23.43 | dnit | Our system ( asterisk) |
14:23.51 | igcewieling | whjy? The calls were not "answered". |
14:24.58 | dnit | But these are out of service numbers right ? |
14:25.22 | dnit | So it would be good if we mark them as OOS and not dial them again. |
14:25.36 | igcewieling | "out of service" can mean "number does not exist", in the USA it usually does. |
14:25.49 | igcewieling | when you call those numbers from other carriers, do they work? |
14:26.17 | dnit | No. Same message in early media on all carriers |
14:26.27 | dnit | Without SIP signals |
14:26.32 | igcewieling | Then the number doesn't exist. |
14:26.53 | dnit | Yes I and you know it but how does the system learns that ? |
14:26.53 | igcewieling | you are using SIP for connection to PSTN, correct? |
14:27.09 | dnit | For asterisk its just a number which has not been answered. |
14:27.37 | dnit | We are using SIP trunking to make all the calls via different carriers |
14:27.40 | igcewieling | dnit: If your carrier isn't giving OOB signaling there is nothing you can do. |
14:29.05 | dnit | Thats when I thought of either making the agents hear the Early media or use https://github.com/phatjmo/app_cpa to process early media |
14:30.21 | igcewieling | your idea is more of a realtime speech recognition. |
14:31.29 | igcewieling | When we call numbers not in service, our carrier sends back a SIP 404 to us. |
14:34.57 | nibbier | I have one problem left with my asterisk installation. it connects via some sip trunk and has internal sip phones. when I connected a specific external incoming call to some other external number asterisk build a simple-bridge, but no sound was transferred. I fixed this by first Playback(somesound) to the caller and then dialing out. This works now. I have the same issue when one of the internal sip phones sens a redirect to an external number when it's |
14:34.57 | nibbier | called. Asterisk builds a bridge but no sound. I could unconditionally playback a sound before Dialing the sip phone, but I'd rather only do this in the external-redirect case. any hints? |
14:35.11 | dnit | In my case they do give 404 for few but for most of such numbers it just the early media unless the call timesout |
14:40.50 | [TK]D-Fender | NIB |
14:41.01 | [TK]D-Fender | nibbier, You're allwing re-invites with your peers. Stop allowing them |
14:42.55 | nibbier | [TK]D-Fender: the redirect itself is the reinvite? |
14:43.55 | [TK]D-Fender | No. What that call goes through allows it |
14:44.01 | [TK]D-Fender | fix your peers <- |
14:44.32 | nibbier | [TK]D-Fender: I'll read up on those reinvites, disable them for my sip phones and try to understand. thanks for the pointer |
14:44.48 | [TK]D-Fender | directmedia=no <-------------- |
14:44.52 | [TK]D-Fender | nothing to read up. |
14:44.59 | [TK]D-Fender | Put it in all of your peers |
14:50.43 | dnit | Can I access early media using prebridge handler ? |
14:52.50 | nibbier | [TK]D-Fender: have directmedia=no in the forwarding sip phones extension. still does not work. |
14:53.10 | samwierema | Is it possible in PJSIP's endpoint configuration to set two (or more) outbound proxies and load balance between them? |
14:54.17 | [TK]D-Fender | it isn't the forwarder |
14:54.17 | [TK]D-Fender | the FORWARDER is the the channel that continues |
14:54.17 | [TK]D-Fender | [TK]D-Fender> Put it in all of your peers <----I was very clear here |
14:54.44 | [TK]D-Fender | dnit, no |
14:55.07 | [TK]D-Fender | dnit, Your idea is DOA. * was not built to accomodate this |
14:55.19 | [TK]D-Fender | samwierema, No. Load balancing is up to you |
14:55.47 | samwierema | [TK]D-Fender: hm, ok. Thanks! |
14:57.30 | nibbier | [TK]D-Fender: I put this in the genear section of my sip.conf... replaced any directmedia=yes by directmedia=no.... no success |
14:58.15 | [TK]D-Fender | ALL OF YOUR PEERS |
15:03.55 | nibbier | sadly I don't even know what a peer is, if you talk about all entities that are defined as type=peer or w/e |
15:05.06 | [TK]D-Fender | every SECTION |
15:05.46 | [TK]D-Fender | regardless of "type=" |
15:05.53 | [TK]D-Fender | AND [general] |
15:09.57 | dnit | [TK]D-Fender: Is there any alternative or I just can't acess ealry media in apps like AMD. |
15:10.23 | [TK]D-Fender | What happens when you send the call there? |
15:10.33 | [TK]D-Fender | What happens when you try to monitor it? |
15:10.39 | [TK]D-Fender | You should already have this answer... |
15:10.40 | igcewieling | dnit: if you want to write your own asterisk application you may do so. |
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15:19.37 | dnit | pre bridge handler is invoked after the call is picked up but before it is bridged ? |
15:19.46 | dnit | ah! |
15:22.03 | dnit | igcewieling: Looks like ill have to do so. Any hint which part of source code I should study first to accomplish this ? |
15:23.42 | igcewieling | dnit: first you'll need to learn things like C, audio processing methods, speech recognition, etc. |
15:24.13 | igcewieling | look at the AMD app for a sample |
15:24.34 | igcewieling | Personally, I think anyone who tries to do that is an idiot. |
15:25.20 | dnit | I wont create my own amd. I just want to use current amd on early media. |
15:25.34 | igcewieling | It is like saying that because my cellphone can't do X, then I'll start a cellphone carrier and have feature X. Yes, that might work. No, it isn't a good idea. |
15:25.42 | dnit | So simply a wrapper on AMD with access to early media will do my work |
15:25.58 | [TK]D-Fender | Again what happens when you TRY the handler NOW? |
15:26.27 | dnit | [TK]D-Fender: It is invoked after the call is picked uip |
15:27.34 | dnit | \help |
15:27.58 | [TK]D-Fender | #asterisk-dev <----- |
15:28.57 | igcewieling | goes back to troubleshooting dialplan variable scope. |
15:30.11 | [TK]D-Fender | igcewieling, Another 2 degrees up and don't forget to account for windage. |
15:33.05 | igcewieling | [TK]D-Fender: and take off the lens cap. 8-) |
15:33.53 | igcewieling | [TK]D-Fender: some sort of race condition between 2 legs of a .call |
15:34.07 | [TK]D-Fender | WHY THE RUSH?! |
15:34.28 | igcewieling | [TK]D-Fender: users! they always want it yesterday. |
15:36.24 | igcewieling | I few months ago I re-wrote part of my dialplan to eliminated the need for chan_local, but since nobody actually *uses* the feature, nobody realized for a few months. |
15:36.44 | igcewieling | nobody actually uses the .call files feature I'm troubleshooting that is. |
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17:05.52 | wabbits | Who do you recommend vm hosting? |
17:11.09 | drmessano | Linode |
17:13.13 | wabbits | thanks drmessano |
17:16.54 | lorsungcu | wabbits: vultr.com |
17:19.03 | wabbits | thanks lorsungcu |
17:50.11 | drmessano | wrong |
17:50.21 | drmessano | He asked for recommendations |
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19:23.16 | Jack17 | Hello guys i have asterisk 11.23.1 im trying to add a trunk that has two ips a signaling ip and media ip any idea guys ? |
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19:25.02 | file | Asterisk isn't configured in regards to the remote media IP, that is part of the negotiation process when doing a call |
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19:51.50 | igcewieling | Jack17: ignore the media ip when setting up the peer |
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20:57.45 | Samot | He didn't have any of his NAT/external IP stuff setup |
20:57.56 | Samot | He asked in #freepbx too since he has FreePBX |
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21:28.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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21:46.24 | FuriousGeorge | hey all |
21:58.14 | FuriousGeorge | could someone kindly take a look at this sip debug output for me. im trying to figure out why im not getting video , but i dont see anything obvious |
21:58.18 | FuriousGeorge | https://pastebin.ca/3842624 |
21:58.42 | FuriousGeorge | unfortunately, i can only debug it on site, and so ium stuck here |
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23:21.23 | Bhakimi | hi guys, how can i adjust a volume of a channel using asterisk AMI ? |
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