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| 04:48.21 | life_of_e | compiling from source when I start asterisk after switching my configs to pjsip.conf res_pjsip does not load and gives this error: undefined symbol: pj_ssl_cipher_name |
| 04:48.59 | life_of_e | res_pjproject did load first |
| 04:52.43 | life_of_e | I compiled using --with-pjproject-bundled |
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| 05:48.50 | life_of_e | eh, figured it out, ssl-dev missing |
| 05:59.03 | life_of_e | so my edits to res_rtp_asterisk.c work, it sends signal 16 and my ATA hook flashes |
| 06:04.04 | life_of_e | hmm, not quite, failed to authenticate |
| 06:05.26 | life_of_e | I don't think I got the pjsip.conf settings correct but I'm not sure what's wrong. I did use the python script to migrate my sip.conf over |
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| 06:12.38 | life_of_e | I'll have to dig more into the pjsip configuration, fix a few things but I did get the ATA to finally dial and the flash worked |
| 06:21.05 | life_of_e | So here's one I don't understand yet. In chan_sip I was able to dial the ATA using SIP/ht813pstn/${EXTEN} where sip.conf contained a port=5062. With chan_pjsip I had to dial using PJSIP/ht813pstn/sip:${EXTEN}@ht813.local:5062. What am I missing that would give me back the previous dialing method? |
| 06:24.03 | drmessano | life_of_e: Do you have a matching peer for ht813? |
| 06:24.25 | life_of_e | yes |
| 06:24.27 | drmessano | You're also showing me two different examples |
| 06:24.51 | life_of_e | I am? |
| 06:25.19 | life_of_e | I'm not following |
| 06:25.22 | drmessano | Are you dialing an exten or a URI? |
| 06:26.15 | life_of_e | I'm guessing a URI because that's the format the ATA needs to dial out originally SIP/device/phonenumber |
| 06:27.34 | life_of_e | I just had a dialplan rule that took any number that matched any 1+10 digit NANP phone number and dialed out |
| 06:27.35 | drmessano | If you have an endpoint matching the ATA you simply need to dial PJSIP/ht813pstn/number |
| 06:28.24 | life_of_e | The ATA is listening on 5062 which is I think part of the problem. How do I inform pjsip of that particular port number for that endpoint? |
| 06:30.09 | life_of_e | Here's the part of pjsip.conf with the ATA's FXO port https://pastebin.com/QtZCHKLW |
| 06:35.13 | drmessano | I guess you can either used a fixed ip:port in the AOR or figure out how to get the ht814 to add the port to the contact when it registers |
| 06:36.29 | drmessano | I'm surprised it's nor already doing thaty |
| 06:37.23 | drmessano | Just out of curiousity |
| 06:37.30 | drmessano | You showed this above |
| 06:37.34 | drmessano | PJSIP/ht813pstn/ |
| 06:37.52 | drmessano | But i see this: |
| 06:37.54 | drmessano | ht8131pstn |
| 06:38.09 | life_of_e | typo |
| 06:38.38 | life_of_e | the hostname is ht813 (which I really should have made it ht8131), the device name within sip/pjsip is ht8131 |
| 06:39.40 | life_of_e | I don't see anything in the ht's config that would block sending the port |
| 06:39.44 | drmessano | what do you get with pjsip show endpoint |
| 06:40.15 | drmessano | what do you get with pjsip show endpoint ht8131pstn |
| 06:40.24 | drmessano | What contact do you have there? |
| 06:40.58 | life_of_e | working on it |
| 06:41.36 | life_of_e | ht8131pstn/sip:ht8131pstn@10.0.50.250:5062 |
| 06:42.57 | drmessano | I can't see why your dial isn't working then |
| 06:43.07 | life_of_e | I'm not sure either |
| 06:43.08 | drmessano | What does the call show |
| 06:45.28 | life_of_e | Endpoint 'ht8131pstn': Could not create dialog to invalid URI 'myphonenumber'. Is endpoint registered and reachable? |
| 06:46.01 | life_of_e | that's with PJSIP/ht8131pstn/myphonenumber as the argument to Dial() |
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| 07:06.15 | life_of_e | I hope I don't have to hardcode the ATA into the dialplan as a URI |
| 07:17.05 | life_of_e | do I need to add a contact to the aor? |
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| 09:15.19 | manolete | Hi, I have an issue where ITSP sends media not negotiated in SDP. Asterisk then Declines the session. Any way I can keep the session and just ignore that media? |
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| 13:13.43 | Samot | life_of_e: it helps not to use Chan_SIP dial strings when calling a PJSIP endpoint |
| 13:14.21 | Samot | PJSIP/${EXTEN}@endpoint |
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| 13:59.30 | life_of_e | Samot I'll try that format later today and see what happens. |
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| 16:56.50 | Janos | hey there, trying to achieve something that initially looked like ConfBridge was the answer but now I'm not so sure. From what I have read from ConfBridge it's great for a full fledge conferencing solution. But what if all I want is to ADD someone to the current call ? Would you guys still recommend ConfBridge or is there a simpler way ?> |
| 16:57.29 | file | besides the DTMF functionality for turning a call into three way, there is no other way |
| 16:58.05 | Janos | where can I find info about this 3-way functionality ? |
| 16:58.22 | Janos | features.conf ? |
| 16:58.35 | file | https://github.com/asterisk/asterisk/blob/master/configs/samples/features.conf.sample#L33 |
| 16:59.24 | Janos | thanks, going to read more about it, looks like it might be just what I need |
| 17:03.18 | Samot | Janos: Add them how? |
| 17:03.58 | Janos | well I was hoping for something similar to a transfer |
| 17:04.42 | Janos | like dialing a DTMF pattern and creating a new call to the other ext and then bridging all the calls where we can all talk and hear each other |
| 17:04.56 | Janos | looks a lot like what @file suggested |
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| 17:09.55 | Samot | OK but most phones do local conferencing. |
| 17:10.05 | Samot | To support 3-way calling or N-way calling. |
| 17:10.55 | Samot | If this is something that is user initiated, as in *I* want to have a 3-way calls between two parties, then I can put a party on hold, call the other and the phone should have a "Join" or "Conference" option to bridge us all together. |
| 17:13.53 | Janos | Samot, sadly the phones in used here are Cisco 7945 IPPhones, while they do have a "Conference" feature it seems to talk a cisco specific crap that asterisk does not understand |
| 17:13.58 | Janos | pardon my french |
| 17:14.18 | Janos | the 7960 does have this join feature and works great |
| 17:14.40 | Janos | but these 7945 don't seem to work with asterisk at all |
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| 17:18.17 | Samot | Sigh. |
| 17:18.30 | Samot | Because those phones were designed for the UCM. |
| 17:18.58 | Samot | Not to be used with other systems. They would require a patch on Asterisk for them to work properly for those features. |
| 17:19.08 | Janos | yeah |
| 17:19.15 | Samot | So yeah, In-Call transfer/3-way are your best bets. |
| 17:19.35 | Janos | kk, will try |
| 17:20.59 | Janos | if you have any doc related to this specific functionality I would appreciate it, the features.conf file is not very clear on how to actually used the atxferthreeway feature |
| 17:21.41 | file | you enable the t or T option to Dial, hit the key to do an attended transfer, then use the 3way key to turn it into a 3way call |
| 17:21.58 | Janos | thanks a lot, will try it |
| 17:50.38 | sibiria | what's the right place to post a bug report these days? |
| 17:51.19 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
| 17:51.36 | sibiria | thanks, file |
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| 19:42.38 | devdvd | Does anyone use the digium a-series phones? if so, how do you like them? is the provisioning fairly straight forward with asterisk? |
| 19:53.06 | Samot | Those a dead phones. |
| 19:53.27 | Samot | The DPMA never supported them. |
| 19:53.58 | Samot | If you're going with Digium phones, S or D series. |
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| 20:01.07 | devdvd | Hmm.. S series? haven't seen anything about them. The D series is out of my price range for what I need. |
| 20:03.53 | Samot | Perhaps that's Sangoma now. |
| 20:04.05 | Samot | D series is pretty much it. A series did not go over well. |
| 20:04.24 | devdvd | What was the issues with them? |
| 20:04.48 | Samot | Not supported in the DPMA, low cost. |
| 20:05.07 | Samot | While supporting the use of a provisioning server, nothing was there to provision it with. |
| 20:05.35 | Samot | I guess there were some TLS issues and some other things. They just didn't last long in regards of being an option. |
| 20:06.09 | devdvd | hmm...digium looks to still sell them? |
| 20:06.10 | Samot | Sangoma was offloading them at almost 75% MSRP. |
| 20:06.42 | Samot | I mean as basic phones, they work. |
| 20:07.03 | Samot | That is what they were designed for. But there's no provisioning management in Switchvox, Asterisk, FreePBX... |
| 20:08.08 | devdvd | and that's fine, i have no issues setting up an ftp server for them to pull the configs from. The only features I really need are kinda basic. BLF, VMI, conference, hold and call transfer |
| 20:09.08 | devdvd | and reading through the docs on the wiki, blf is alot cleaner to setup on these than on the polycom sp650 that im testing |
| 20:11.58 | Samot | Well in all fairness, the SP's are EOL'd. |
| 20:12.06 | devdvd | this is very true |
| 20:12.11 | Samot | The VVX series is the current series. |
| 20:12.56 | Samot | But also understand the A-Series is done. I highly doubt it will get any fixes or updates. |
| 20:13.16 | devdvd | interesting |
| 20:13.33 | devdvd | what's the replacement? |
| 20:13.40 | Samot | There isn't any. |
| 20:13.52 | Samot | It was an experiment from what I can tell. |
| 20:14.03 | Samot | A low cost, low impact phone as an option. |
| 20:14.09 | Samot | It didn't replace the D series. |
| 20:14.42 | devdvd | of course, so other than them not having DPMA support, what other issues should I look out for? |
| 20:14.43 | Samot | I mean it had an actual life span of under 12 months or so. |
| 20:14.55 | Samot | I'm not sure. |
| 20:14.58 | devdvd | then why are they still selling them? |
| 20:15.10 | Samot | They were launch in early 2018 and stopped by the end of 2018. |
| 20:15.12 | devdvd | im not doubting you, i'm just questioning them |
| 20:15.23 | Samot | Because they have a metric shit tonne of stock. |
| 20:15.34 | Samot | They are "as-is" phones from what I can tell at this point. |
| 20:16.33 | Samot | I know there was a TLS issue because someone bought like 20 at the 75% off price... |
| 20:16.46 | Samot | Didn't test them and put them in the field with TLS and had nothing but issues |
| 20:16.55 | Samot | Turn off TLS, fine. Turn it on, all sorts of issues. |
| 20:17.09 | Samot | Had to replace phones that needed TLS support with Yealinks. |
| 20:17.37 | Samot | I believe he was told no further firmware updates were forthcoming. |
| 20:19.01 | Samot | 4:15:00 PM <devdvd> then why are they still selling them? << The honest honest, consumers like yourself who may not mind they are "as-is" |
| 20:19.12 | Samot | s/honest/answer/ |
| 20:20.11 | Samot | When you're a business that invested X amount of dollars in a product and it ends up failing. You're still sitting on Y amount of product collecting dust. That's a complete loss. |
| 20:20.42 | devdvd | i understand that |
| 20:22.00 | Samot | They are like what? $75-90 phones normally. Sangoma was/is selling them at $29.99. |
| 20:22.08 | Samot | And were giving deals on bulk. |
| 20:22.34 | Samot | That's pretty much cost if not below cost. |
| 20:24.02 | devdvd | hmm...not what i found |
| 20:24.40 | devdvd | Looking at the A25's, cheapest i found was 109 each on amazon, digium still has them at like 139 |
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| 20:29.14 | devdvd | Thank you for the feedback Samot, I appreciate it. |
| 20:29.40 | Samot | If you have a FreePBX box and an account in the portal, look at the Sangoma portal. |
| 20:29.46 | Samot | They still might have them cheap. |
| 20:31.26 | Samot | A25's are $60 through Sangoma. |
| 20:32.15 | Samot | Looks like the current range is $40 to $85 for the A Series with Sangoma. |
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| 20:44.06 | devdvd | ah interesting, ok ill check that out |
| 20:44.28 | devdvd | where'd you find that at? |
| 20:48.09 | devdvd | ah i see what you said. I just created an account on the freepbx site, i hvae a freepbx box |
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| 21:53.15 | KNERD | Well, I completed the task of using the converter |
| 21:53.41 | KNERD | I switched all the extensions to PJSIP and now all the extensions keep going on an offline |
| 21:56.12 | KNERD | any ideas why this would be goign on? |
| 21:56.18 | Samot | Which converter? |
| 21:56.28 | KNERD | the one you mentioned |
| 21:57.56 | KNERD | http://convert.freepbx.org/ |
| 21:57.56 | Samot | And then you converted the extensions from Chan_SIP to Chan_PJSIP? |
| 21:57.56 | KNERD | yeah since PJSIP was already listening on port 5060 |
| 21:58.17 | Samot | And did you go into the Asterisk SIP Settings and setup the Chan_PJSIP stuff? |
| 21:58.21 | Samot | Local networks, etc? |
| 21:58.30 | KNERD | yes |
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| 21:58.47 | Samot | And are the extensions local to the PBX? |
| 21:58.52 | KNERD | no |
| 21:59.24 | Samot | asterisk -rx "pjsip show transport 0.0.0.0-udp" |
| 21:59.28 | Samot | Pastebin it |
| 22:00.15 | KNERD | https://pastebin.com/1L4B4AFz |
| 22:00.39 | Samot | OK, sigh. |
| 22:00.51 | Samot | Dude if the box is remote then there are no local networks. |
| 22:01.20 | Samot | Everything is remote. |
| 22:01.32 | Samot | Does the box have a public IP? |
| 22:01.50 | KNERD | Yes. |
| 22:01.56 | Samot | 216.218.249.181 <-- Is that the actual IP of the box? |
| 22:02.07 | KNERD | I just clicked on Deteck Network sSettings, sdont shoot the messnger |
| 22:02.11 | KNERD | yes |
| 22:02.38 | Samot | So remove all the local network stuff |
| 22:03.21 | KNERD | In the olden days there was a choice of thr type of Ip address |
| 22:03.45 | Samot | In the olden days there was only Chan_SIP |
| 22:03.55 | Samot | PJSIP is a much more robust SIP driver. |
| 22:04.40 | KNERD | Okay done, and reloaded, but they are still doing it |
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| 22:05.25 | Samot | What are they still doing? |
| 22:05.28 | Samot | What is the actual issue? |
| 22:05.45 | KNERD | the same thing I was complainign about. the extensiosn keep goign off and online |
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| 22:06.13 | file | what does that actually mean though? |
| 22:06.34 | KNERD | Here |
| 22:07.07 | KNERD | <PROTECTED> |
| 22:07.09 | KNERD | <PROTECTED> |
| 22:07.11 | KNERD | <PROTECTED> |
| 22:07.13 | KNERD | <PROTECTED> |
| 22:08.08 | file | that occurs as a result of PJSIP sending a SIP request and getting no response, you may need to increase the rate at which it sends those SIP requests to ensure the NAT Mapping stays open |
| 22:08.09 | Samot | That's still not showing us anything. |
| 22:08.27 | Samot | What is the qualify frequency set to? |
| 22:08.34 | Samot | On the extensions.. |
| 22:09.07 | KNERD | it shows 25 |
| 22:12.03 | Samot | And what kind of router is in front of the phones? |
| 22:12.16 | *** part/#asterisk kharwell (kharwell@nat/digium/x-pgbcgotrpwtslmek) |
| 22:12.53 | KNERD | Ebiquity ER-X |
| 22:13.10 | Samot | Well this is a NAT issue it's looking like. |
| 22:13.50 | Samot | If the PBX is sending keep-alive every 25 seconds and your phones can't response within 7 seconds there is an issue with the router/firewall |
| 22:13.54 | Samot | Or something |
| 22:14.14 | KNERD | i changed some extensions to 10 and still going offline |
| 22:14.32 | KNERD | I guess I will have to swap back over to chan_sip |
| 22:15.08 | Samot | 6:08:10 PM <@file> that occurs as a result of PJSIP sending a SIP request and getting no response, you may need to increase the rate at which it sends those SIP requests to ensure the NAT Mapping stays open |
| 22:15.27 | KNERD | yeah I saw that |
| 22:15.29 | Samot | The PBX is sending requests to the phones and not getting a response.... |
| 22:15.35 | Samot | But did you understand it? |
| 22:15.44 | KNERD | of course |
| 22:15.58 | KNERD | there was no problem with chan_sip for past two years or so |
| 22:16.10 | Samot | OK, switch back. |
| 22:29.59 | KNERD | well this pBX has serious issues |
| 22:30.33 | *** join/#asterisk dimestop (~h@2607:5300:201:3000::16d5) |
| 22:30.54 | KNERD | I switch to pjsip only and it's listening on port 5060, but when I so "sip show peers' the only thing which appears is the trunk, no extensions |
| 22:31.42 | dimestop | I have a problem with the ami originate command being delayed by 20-30 seconds at around 100 calls.. can anyone lend a hand? |
| 22:32.25 | dimestop | KNERD: pjsip show channels |
| 22:32.31 | dimestop | sip/pjsip have different commands |
| 22:32.37 | dimestop | *chansip/pjsip |
| 22:32.44 | KNERD | i am not using pjsip |
| 22:32.50 | dimestop | "I switch to pjsip only" |
| 22:32.52 | dimestop | what did you mean |
| 22:32.59 | KNERD | sorry, I meant chan_sip |
| 22:33.02 | dimestop | oh okay |
| 22:39.58 | KNERD | Peronsally I think this is an issue with running the Sangoma OS on KVM |
| 22:40.25 | *** join/#asterisk samwierema (~samwierem@195.240.143.134) |
| 22:53.09 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:57.45 | dimestop | I have confirmed my issue by doing a pcap on the ami port. I can see the originate request and then the agent is not dialled for 20-30 seconds later |
| 22:57.56 | dimestop | this is a big problem.can anyone help? |
| 22:58.02 | dimestop | asterisk 13.19.1 |
| 23:05.30 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 23:08.23 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 23:08.25 | KNERD | dimestop: running in a virtual machine? |
| 23:12.40 | Samot | dimestop: You need to be more descriptive of the issue. |
| 23:13.24 | Samot | You're sending an Originate via AMI? |
| 23:13.27 | KNERD | Yeah, something is VERY wrong with running Sangoma OS in KVM, Now out of the bluie, asterisk is reporting "No such command 'pjsip show enpoints' (type 'core show help pjsip show enpoints' for other possible commands)" |
| 23:13.38 | Samot | No there is not. |
| 23:14.22 | KNERD | yeah, that was a typo |
| 23:14.22 | Samot | I run everything 100% in KVM. |
| 23:14.22 | Samot | Yes, it was. |
| 23:14.22 | KNERD | then why nothig is showup up for chan_sip when there was 10 extensions for chan_sip? |
| 23:14.24 | KNERD | it should show something, even if offline |
| 23:14.28 | Samot | Did you do sip show peers? |
| 23:14.42 | Samot | Chan_SIP and Chan_PJSIP have different commands, that should be obvious. |
| 23:14.52 | KNERD | yes, the only thing showing up is the SIP trunk |
| 23:15.36 | KNERD | 20 extensiosn and I now have 10 on chan_sip and the other 10 on PJSIP |
| 23:16.15 | drmessano | core reload |
| 23:16.59 | KNERD | i have done more than that, rebooted the machine twice |
| 23:17.47 | drmessano | module show like sip |
| 23:17.58 | dimestop | I'll get some more info for you |
| 23:18.00 | Samot | asterisk -rx "sip show peers" |
| 23:19.10 | KNERD | here module show like sip https://pastebin.com/3Yz6FaUv |
| 23:19.12 | dimestop | it is a virtual and we are sending via ami, but after we see the request in the pcap.. it takes another 20-30 seconds to actually dial |
| 23:19.19 | dimestop | Samot/KNERD |
| 23:19.40 | KNERD | See, I knew it..he was running in virtual machine |
| 23:19.53 | Samot | So? |
| 23:19.58 | Samot | Again, I do everything in VMs. |
| 23:20.00 | KNERD | that is whatissue I was having. the old system got to running very very slow |
| 23:20.34 | KNERD | would take 51 seconds for a call to go out |
| 23:20.36 | Samot | dimestop: What did you see when you debugged Asterisk Manager? |
| 23:21.20 | Samot | KNERD: Please stop relating unrelated things. |
| 23:21.20 | drmessano | You probably ran out of disk space⦠LOL |
| 23:21.20 | KNERD | no, there was plenty |
| 23:21.21 | Samot | dimestop has yet to provide any real information. |
| 23:21.47 | drmessano | By the way, I run all my PBX instances in KVM as well |
| 23:21.56 | drmessano | Literally not a single issue |
| 23:21.59 | Samot | KNERD: 7:18:01 PM <Samot> asterisk -rx "sip show peers" |
| 23:22.50 | KNERD | As I mentioned it only shows the trunk |
| 23:23.04 | dimestop | sorry I'm working on it live with a colleague |
| 23:23.14 | dimestop | I'll get back to you in a minute, thanks for the help though |
| 23:23.17 | KNERD | even though there are now 10 extensions set to chan_sip |
| 23:24.09 | Samot | asterisk -rx "sip show settings" |
| 23:24.11 | Samot | Do that then. |
| 23:25.11 | KNERD | https://pastebin.com/4ZkZUBiv |
| 23:26.08 | Samot | Did you change the PJSIP port as well? |
| 23:26.45 | Samot | They both can't be on 5060 |
| 23:27.12 | Samot | Please tell me you haven't been using both on 5060 this entire time... |
| 23:27.15 | KNERD | of course I did |
| 23:27.18 | Samot | OK. |
| 23:27.42 | Samot | asterisk -rx "sip show peer <extension>" |
| 23:27.55 | Samot | Replace <extension> with one of the ones you converted. |
| 23:30.36 | KNERD | it returns "not found" |
| 23:31.19 | KNERD | though the extension listing shows it as sip |
| 23:31.40 | Samot | asterisk -rx "pjsip show transport 0.0.0.0-udp" |
| 23:33.32 | KNERD | https://pastebin.com/QDUnCgR0 |
| 23:34.35 | Samot | Create a new Chan_SIP extension, see what happens. |
| 23:36.05 | KNERD | okay, it is created |
| 23:36.17 | KNERD | and shows up with sip show peer 400 |
| 23:36.39 | KNERD | and sip show peers |
| 23:36.41 | Samot | So then something has happened. |
| 23:36.47 | Samot | It didn't convert right or something. |
| 23:37.31 | KNERD | yeah |
| 23:37.45 | KNERD | I guess I will just install this on a physical machine |
| 23:38.04 | KNERD | I have had two other PBXs on KVM acting very flaky |
| 23:38.13 | KNERD | thanks for assist |
| 23:38.29 | dimestop | Samot: what do you mean by debugged asterisk manager? |
| 23:41.22 | dimestop | if there's a way to debug further than pcap on the ami port, then that would definately be helpful |
| 23:42.37 | drmessano | #MaRcHsAdNeSs |
| 23:44.05 | drmessano | dimestop: there's a manager debug command in the CLI |
| 23:46.08 | dimestop | looks like this is the same stuff that was shown in the pcap |
| 23:46.19 | dimestop | so in other words, we get the originate |
| 23:46.22 | dimestop | and then 30 seconds later |
| 23:46.26 | dimestop | it starts doing the dial begin |
| 23:46.49 | dimestop | this seems to only happen when asterisk has over 100 calls |
| 23:46.59 | drmessano | Are you using chan_sip? |
| 23:47.04 | dimestop | yes |
| 23:47.26 | dimestop | 8 core CPU, load average is around 4-5 |
| 23:47.32 | dimestop | at its peak |
| 23:47.46 | drmessano | Do you have a local DNS resolver? |
| 23:47.46 | dimestop | its around 9 |
| 23:48.18 | dimestop | I'll double check that |
| 23:49.05 | dimestop | yes it does |
| 23:49.11 | dimestop | its hosted in AWS |
| 23:49.19 | dimestop | if that matters at all |
| 23:51.36 | drmessano | Yeah there's latency somewhere else but can't say without seeing core debug I guess. I'm mobile tho |
| 23:51.56 | dimestop | I'll stop some other processes and see if a lighter load helps |
| 23:52.40 | Samot | Oh, AWS. |
| 23:53.25 | Samot | The EBS is a POS when it calls to real call volumes. |
| 23:54.12 | dimestop | interesting |
| 23:54.15 | dimestop | I'll pass that along |
| 23:54.16 | dimestop | thanks |
| 23:55.10 | Samot | You're making I/O requests to a NAS basically. |
| 23:55.28 | drmessano | AWS is basically the worst platform for hosted voice |
| 23:55.35 | Samot | True dat. |
| 23:56.28 | dimestop | It has a 600 IOPS on the main drive, recordings are on a 2000 IOPS volume |
| 23:58.01 | drmessano | And you're recording calls? |
| 23:59.00 | dimestop | yes, all calls are recorded to the high iops volume |
| 23:59.39 | drmessano | You must hate your life lol |