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| 00:18.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12) 16.0.0-rc1 (2018/08/08), Standard: 15.5.0 (2018/07/12); 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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| 04:59.26 | wyoung | .o/ |
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| 06:00.32 | Apocryphal | Is there some way to force asterisk to use simple_bridge for a call instead of native_rtp? |
| 06:01.50 | Apocryphal | Selectively - that is. Asterisk 13 seems to have some habit of selecting native_rtp with PCMA in one end, and PCMU in the other, which cause one-way audio |
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| 11:48.56 | BlackBishop | so when I try to run asterisk -vvvvdddd I get one of this messages |
| 11:49.23 | BlackBishop | it only happens when I have res_odbc.conf configured correctly |
| 11:55.54 | BlackBishop | here's more info and content .. http://dpaste.com/0CZJ7TN |
| 12:21.05 | BlackBishop | recompiled dev-db/myodbc dev-db/mysql-connector-c dev-db/mysql-connector-c++ dev-db/unixODBC net-misc/asterisk .. now I get 3 new errors ...! :D http://dpaste.com/3KTGQTG |
| 12:22.58 | Samot | What are you trying to do? |
| 12:23.44 | BlackBishop | start asterisk and get to the cli... |
| 12:24.12 | Samot | What version and what OS? |
| 12:25.14 | sibiria | obvious ABI breakage there. did you compile these things on two different OSes or something? |
| 12:26.30 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 12:27.09 | BlackBishop | sibiria: nope :| |
| 12:27.19 | BlackBishop | without the res_odbc there works ! :| |
| 12:27.41 | BlackBishop | Samot: http://dpaste.com/0CZJ7TN 13.22.0 |
| 12:30.15 | Samot | OS/? |
| 12:30.18 | BlackBishop | gentoo |
| 12:30.20 | BlackBishop | :| |
| 12:30.26 | Samot | Oh |
| 12:30.28 | BlackBishop | but everything else in the OS works just fine for years now :| |
| 12:31.05 | BlackBishop | had no issues .. I had this (asterisk) setup working a couple of years ago .. and I wanted to use it again, so I upgraded the required packages but I'm missing something |
| 12:32.12 | BlackBishop | everything else on the machine is great .. apache,postfix,dovecot,sql,docker,openvpn,grafana,hbase,kodi,virtualbox,bind,ntp and so on... |
| 12:32.50 | BlackBishop | I use this machine for testing but it's been stable for years... |
| 12:33.27 | Samot | Im sorry...Kodi? |
| 12:33.28 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 12:33.39 | BlackBishop | Samot: https://kodi.tv/ :)) |
| 12:33.45 | Samot | Youre running streaming apps on your pbx? |
| 12:33.53 | Samot | I know what it is |
| 12:34.03 | sibiria | it looks more like a big general entertainment/utility machine more like a PBX |
| 12:34.05 | BlackBishop | like I said, it's a testing environment |
| 12:34.08 | BlackBishop | It's the other way around |
| 12:34.10 | Samot | Ok |
| 12:34.25 | BlackBishop | I'm running a pbx on my entertainment machine :)) |
| 12:34.35 | Samot | Sigh |
| 12:34.51 | BlackBishop | but it's ok. it's nothing fancy. everything is personal use. |
| 12:35.46 | Samot | So your production pbx is also an entertainment box? |
| 12:35.54 | BlackBishop | it's not production :) it's testing. |
| 12:36.10 | BlackBishop | it's also used for browsing .. image editing and whatnot :) |
| 12:36.19 | Samot | Sp that means you have a production box? |
| 12:36.31 | BlackBishop | yep. amazon dedicated machine. |
| 12:36.43 | BlackBishop | different distro ( centos 7 ) |
| 12:36.53 | Samot | So your test environment doesnt match your production |
| 12:37.06 | Samot | Thats a poor test environment |
| 12:37.07 | BlackBishop | yeah, but I doubt that matters in any way |
| 12:37.13 | Samot | Ok |
| 12:37.16 | BlackBishop | all I'm using from asterisk is sip and some simple peers |
| 12:37.24 | BlackBishop | inside asterisk itself |
| 12:37.29 | BlackBishop | no outside peering or anything |
| 12:37.43 | BlackBishop | me and a couple of friends use csipsimple to play |
| 12:37.51 | Samot | Im not sure what that means |
| 12:38.02 | Samot | "No outside peering" |
| 12:38.25 | BlackBishop | there's no calling to other lines ( land or mobile ) in or out |
| 12:38.37 | Samot | Ok |
| 12:38.41 | BlackBishop | through any 3rd party |
| 12:38.58 | Apocryphal | I'm hoping someone can offer some sort of insight. I've got a SIP device that is misbehaving some. Even when it's holding it's peer, it still sends RTP traffic. In a native_rtp bridge, this seems to completely mess up the hold music being played to the peer being held. This doesn't happen if the bridge is simple_bridge |
| 12:39.02 | Samot | Your phones are peers. |
| 12:39.25 | BlackBishop | my bad, so we're only using csipsimple peers to call one another through the box |
| 12:39.45 | BlackBishop | and now I wanted to add some voicemail and wanted to "devel" on my box |
| 12:40.01 | BlackBishop | but this fails because of res_odbc |
| 12:40.12 | BlackBishop | which is used for the peers details |
| 12:41.26 | BlackBishop | that box is in my lan behind a firewalled router ... not that it has much to firewall, only does outgoing nat |
| 12:42.34 | Samot | Your test box should match your production box |
| 12:42.36 | Samot | Period. |
| 12:42.37 | BlackBishop | so, how would I debug what's wrong ? .. recompiling asterisk doesn't fix it :/ and it only happens when the res_odbc module has a working res_odbc.conf |
| 12:42.54 | Samot | Because whatever you do to make this work on this test box isn't going to be the same on the production box. |
| 12:42.56 | Samot | At all. |
| 12:42.57 | Samot | So... |
| 12:43.01 | BlackBishop | ok, I'll delete the production box so consider everything as testing |
| 12:43.12 | BlackBishop | and I'll make a prod box afterwards and see how that works. |
| 12:43.21 | Samot | And is the production box going to be Gentoo? |
| 12:43.27 | BlackBishop | sure, why not ? |
| 12:43.43 | BlackBishop | it's going to be minimal anyway so it shouldn't take too long. |
| 12:43.43 | Samot | I think you're missing my point. |
| 12:43.44 | file | Gentoo is not a distribution that many people use with Asterisk, and if your problem is the ODBC library land - then that's where you're going to have to look... |
| 12:43.52 | Samot | ^^^ |
| 12:44.02 | Samot | Because this is sounding like Gentoo's handling of ODBC |
| 12:44.08 | BlackBishop | file: I agree, but the weird thing is that if I have the wrong password in res_odbc.conf I get to the cli ! |
| 12:44.15 | BlackBishop | if the right password is there, I don't ! |
| 12:44.19 | Samot | Either you didn't download all the dev packages or there's another issue. |
| 12:44.21 | file | that's not weird because it changes the code that is executed |
| 12:45.00 | BlackBishop | Samot: :) everything is source compiled so the dev files are all there... |
| 12:45.18 | Samot | So you say |
| 12:45.25 | file | disregarding Asterisk, there are ways of verifying the ODBC configuration using tools against it and such |
| 12:45.26 | Samot | So you compiled it all? |
| 12:45.30 | file | running queries, etc |
| 12:45.31 | BlackBishop | yep |
| 12:45.34 | Samot | What file said. |
| 12:45.35 | BlackBishop | file: isql is ok |
| 12:45.59 | BlackBishop | # isql -v dex_kyesms says "Connected" |
| 12:46.06 | file | then I don't know, but I don't believe it's code in Asterisk |
| 12:46.20 | BlackBishop | doing a "SELECT * FROM ast_sipfriends" works |
| 12:46.44 | BlackBishop | but res_odbc.so ( which is in asterisk ) doesn't like something |
| 12:46.55 | file | unless however Gentoo has things built or however you've configured somehow requires code changes or other things |
| 12:47.44 | file | but I have no guide or suggestion on how to isolate or figure it out, determining ODBC stuff isn't that simplle |
| 12:47.53 | BlackBishop | :-( |
| 12:48.33 | BlackBishop | even if there's a gentoo bug, the guys there will probably want to know "where" and right now all I can say is that asterisk's res_odbc.so fails for some reason |
| 12:49.01 | BlackBishop | can't even get a good stacktrace .. it's failing with all sorts of messages :| |
| 12:51.26 | Samot | You didn't have this issue on CentOS? |
| 12:51.47 | BlackBishop | no |
| 12:51.56 | Samot | Then I guess you have a choice.. |
| 12:51.56 | BlackBishop | but then again, I didn't have this issue on gentoo last year either. |
| 12:52.01 | file | we have automated tests nightly I think that do CentOS for ODBC + Postgresql |
| 12:52.08 | Samot | Run an OS that is supported and widely used by Asterisk users... |
| 12:52.16 | Samot | Or keep doing this... |
| 12:52.30 | BlackBishop | yeah, there is a bug somewhere .. I'm just curious where |
| 12:52.38 | BlackBishop | maybe there's an incompatible library version |
| 12:54.16 | Apocryphal | Anyone know if there is some dialflag og channel variable or anything I can set, to force a call not to bridge into native_rtp, and use simple_bridge instead? |
| 12:55.02 | file | Apocryphal: the only thing that controls that are enabled features, for example MixMonitor would force a simple bridge |
| 12:56.59 | Apocryphal | Yeah right now I enable monitor to force it - though that records the call so I need to write something that deals with deleting the recordings we don't actually want to record :P |
| 12:57.03 | Apocryphal | But if that's the only way |
| 12:57.08 | Apocryphal | Oh well |
| 12:58.11 | file | there's no options explicitly for controlling it, because it's an implementation detail that shouldn't matter - it should choose appropriately, just in your case the endpoint goes against the expectations... |
| 12:58.22 | Apocryphal | I might as well just disable the bridge_native_rtp module then. My main issue is that some of our phones send RTP even when they're holding a call, and this seems to destroy hold music when native_rtp is briding |
| 12:58.56 | file | you can file an issue as I think it's reasonable for us to deal with that, but I don't have a timeframe on when it would get looked at |
| 12:59.10 | Apocryphal | Yaeh I'll report that as a bug for native_rtp |
| 13:00.03 | file | what version of Asterisk? |
| 13:00.28 | Apocryphal | Right now 13.6, but I've just tested 13.22 and it's present there as well |
| 13:00.36 | file | k |
| 13:02.52 | *** join/#asterisk scgm11_ (~scgm11@r186-49-55-9.dialup.adsl.anteldata.net.uy) |
| 13:09.46 | Samot | Apocryphal: Question on the phones sending RTP.... |
| 13:09.55 | Apocryphal | Samot: yup? |
| 13:09.59 | Samot | Is the RTP showing a 0.0.0.0 for the IP? |
| 13:10.12 | Samot | Or a sendonly option in the SDP body? |
| 13:10.25 | Apocryphal | a=sendonly in the SDP body |
| 13:10.29 | Samot | OK |
| 13:10.31 | Samot | That's common. |
| 13:10.37 | Samot | My Polycoms do that. |
| 13:10.55 | Apocryphal | Hmm |
| 13:11.05 | Samot | Most phones do, some like the Cisco SPAs (that I've seen) just send the IP as 0.0.0.0 |
| 13:11.15 | Samot | So incoming audio isn't received to the phone. |
| 13:11.22 | Apocryphal | Well, the devices sending RTP are JsSIP based WebRTC phones. There is an RTP Engine between asterisk and the browser |
| 13:11.47 | Samot | Well that narrows it down. |
| 13:11.55 | Apocryphal | *something* is interfering with the native_rtp bridge for sure though. Question is exactly what |
| 13:12.06 | Samot | You just said there's another RTP Engine |
| 13:12.13 | Samot | Have you looked there? |
| 13:12.20 | Apocryphal | What would I be looking for though? |
| 13:12.34 | Apocryphal | RTPEngine only sends something if it receives something |
| 13:12.59 | Samot | Wait, you mean RTPEngine like with Kamailio/OpenSIP? |
| 13:13.07 | Apocryphal | Yes |
| 13:13.16 | Samot | OK, then I'd look there.. |
| 13:13.22 | Samot | At the configs.. |
| 13:13.30 | Samot | Or remove Kamailio from the testing... |
| 13:14.00 | Apocryphal | What for though? I'm positive it happens because RTP gets sent - here is why (typing) |
| 13:14.05 | Samot | To claim this is an Asterisk bug or possible one with something in the middle that is designed to change the SDP yeah... |
| 13:14.09 | Samot | It needs to go. |
| 13:14.15 | Samot | So that's it's client/asterisk only |
| 13:14.25 | Samot | If the issue still persists then it could be an Asterisk bug |
| 13:14.30 | Apocryphal | 1) If I yank the Ethernet Cable from the device while it's holding, then hold music resumes and works immediately, and is broken again when the ethernet cable is reattached |
| 13:14.38 | Samot | But you have something in the flow that modifies how RTP is handled. |
| 13:14.43 | Apocryphal | 2) if I, via javascript, destroy the RTPSender on holding, it works as well |
| 13:14.55 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:15.11 | Samot | You have to eliminate anything Kamailio/OpenSIP is doing with RTP |
| 13:15.17 | Apocryphal | Unless it's expected that bridge_native_rtp breaks holdmusic if the holding party sends RTP |
| 13:15.25 | Samot | So that it's a pure sample between the endpoint and Asterisk. |
| 13:16.20 | Apocryphal | Asterisk won't be able to handle the RTP and SIP directly. the proxies are there to translate between the two |
| 13:16.25 | Apocryphal | DTLS vs. SRTP for instance |
| 13:16.30 | Apocryphal | Asterisk can't do both at the same time |
| 13:16.45 | Apocryphal | So RTPEngine converts from DTLS -> SRTP and vice versa |
| 13:17.04 | Samot | Obviously this is from how Asterisk is RECEIVING the SDP body |
| 13:17.09 | file | pedantic: it's SRTP regardless |
| 13:17.19 | file | DTLS is used for key exchange to derive the SRTP keys |
| 13:17.27 | Samot | So you have to look at what is SENDING to Asterisk. |
| 13:17.44 | Apocryphal | Yes, but regadless of pedantism, asterisk can't be configured to support "SRTP" and "DTLS" at the same time |
| 13:18.04 | file | I assume you are actually referring to SDES |
| 13:18.09 | Apocryphal | Samot: As in the issue might be with the SDP? |
| 13:18.16 | Apocryphal | SDES DTLS, sorry yes |
| 13:18.18 | *** join/#asterisk gruetzkopf (gruetzkopf@bnc.dont-follow.us) |
| 13:18.24 | Samot | Show us a call this is happening with |
| 13:18.32 | Samot | Full SIP Debug on Asterisk's side.. |
| 13:18.38 | Samot | Or PJSIP logger... |
| 13:18.45 | Apocryphal | Alright, I'll generate that later today |
| 13:18.48 | Apocryphal | It's chan_sip |
| 13:18.48 | Samot | Let's see these SIP messages for hold/unhold |
| 13:19.11 | *** join/#asterisk gruetzkopf (gruetzkopf@bnc.dont-follow.us) |
| 13:19.22 | Samot | Well I have no guarantees on my availability to help with this "later today" |
| 13:19.24 | Samot | So.... |
| 13:19.42 | Apocryphal | It's not more urgent than we can continue this at a later date :) |
| 13:20.24 | Apocryphal | I can get the SIP INVITES that holds the call in a minute though |
| 13:20.42 | Samot | Yes. |
| 13:20.48 | Samot | A Hold generates an INVITE |
| 13:20.54 | Samot | That's what I was talking about.. |
| 13:20.57 | Samot | sip set debug on |
| 13:21.05 | Samot | make call, put on hold, take off hold... |
| 13:21.09 | Apocryphal | I can pull that from HOMER, hold on |
| 13:21.14 | Samot | No. |
| 13:21.22 | Samot | GD. |
| 13:21.24 | Samot | FROM ASTERISK |
| 13:21.26 | Apocryphal | you want it specifically from sip set debug on |
| 13:21.28 | Apocryphal | Alright |
| 13:22.49 | Samot | pastebin it please.... |
| 13:27.17 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 13:27.36 | Apocryphal | https://pastebin.com/AEKGCRgD |
| 13:28.22 | Apocryphal | Line 380 - that's where a hold invite comes in |
| 13:31.24 | Samot | So when the caller is put on hold, they do not get hold music? |
| 13:32.09 | Apocryphal | They get, but it's extremely distorted (sounds like package loss) |
| 13:32.21 | Apocryphal | The issue disappars if I yank the ethernet cable so the device can't send RTP |
| 13:32.41 | Apocryphal | And it reappears immedaitely if I reattach the cable |
| 13:34.22 | Samot | OK, is this for both inbound/outbound calls? |
| 13:34.46 | Apocryphal | Yes, doesn't matter if the "defective" device is the caller or the callee |
| 13:35.21 | Samot | So the call bridges up fine and you chit-chat.. |
| 13:35.34 | Apocryphal | Yup, when talking there is no issue at all |
| 13:35.39 | Samot | But the moment you put them on hold, they get garbled MoH.. |
| 13:35.44 | Apocryphal | Yup |
| 13:36.07 | Apocryphal | And if I disconnect the defective device from the network (so it can't send RTP while holding), the remote party hears hold music correctly immediately. |
| 13:36.21 | Apocryphal | And the moment I reconnect the ethernet cable, it becomes garbled again |
| 13:36.39 | Samot | And does this happen with other types of endpoints |
| 13:36.48 | Samot | IP phones, other softphones? |
| 13:36.58 | Samot | Other WebRTC clients? |
| 13:37.34 | Apocryphal | The "defective" device is a Softphone - i haven't tried other WebRTC clients. Doesn't happen with any other devices we use (IP Phones, Bria, X-lite etc) |
| 13:38.13 | Samot | I would widen your WebRTC/WSS testing. |
| 13:38.16 | Apocryphal | Sorry, the defective device is a "JsSIP based WebRTC Phone running in Chrome" |
| 13:38.31 | Samot | I'd try another browser.. |
| 13:38.41 | Samot | Another WebRTC phone... |
| 13:39.06 | Apocryphal | Yes, finding a workaround isn't really the issue. I've disabled native_rtp. But more if this is a bug in native_rtp |
| 13:39.12 | Apocryphal | And if it's worth reporting |
| 13:39.15 | Samot | OK |
| 13:39.29 | Samot | So how has this proven to be an Asterisk bug? |
| 13:40.53 | Apocryphal | Well clearly a sip device is able to sabotage something inside native_rtp? as proven by disconnecting the device solving the issue, and reappearing when reconnecting it. Asterisk recognizes the sendonly INVITE and starts playing MOH, but RTP packages being sent to asterisk disrupts the MOH |
| 13:41.01 | Apocryphal | Is that how it's supposed to work? |
| 13:41.33 | Samot | But you can't reproduce it with the other SIP devices that do the EXACT same process. |
| 13:41.45 | Samot | Everything else *works* |
| 13:41.47 | Apocryphal | I don't have any sip devices that sends RTP while holding |
| 13:41.48 | Samot | You just said that. |
| 13:42.20 | Samot | Polycom's tell Asterisk how to hold a call the same way this is happening |
| 13:42.37 | Apocryphal | I'll have to find a polycom device then |
| 13:42.44 | Samot | Other devices as well |
| 13:42.49 | Samot | It's how It is done... |
| 13:43.03 | Samot | so far this is an issue with JsSIP on Chrome. |
| 13:43.11 | Apocryphal | So it's working as intended that the content of the RTP packages can disrupt MOH in a native_rtp bridge? |
| 13:43.13 | Samot | Now it could be related to wss |
| 13:43.32 | Samot | But in order to see if that is the case, you need another wss client |
| 13:44.17 | Samot | Why is the SIP endpoint sending RTP over a line it has held. |
| 13:44.48 | Apocryphal | No idea. What goes on inside JsSIP and Chrome is beyond me unfortunately |
| 13:46.19 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:49.43 | Apocryphal | The MOH issue happens with firefox as well. |
| 13:49.55 | Apocryphal | For what it's worth |
| 13:57.35 | *** join/#asterisk mahafyi (~chatzilla@122.178.79.3) |
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| 14:37.22 | igcewieling | Does anyone know if the chan_sip setting "keepalive" works/applies when using TCP instead of UDP. |
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| 18:39.04 | borninbronx | hi! I'm not using asterisk but I've an issue configuring an hardware device acting as voip-ata adapter regarding the caller ID. I'd appreciate if you can give me some link to understand the subject better |
| 18:39.27 | borninbronx | I'll describe the problem: calls works but I've a problem with the caller ID showing up on my analogic phones. Weirdly enough the issue only happens from landline calls I receive: say a landline is calling from number 01234567, what I receive as caller ID is 3901234567: Italian international prefix is +39 but this is missing the + in front, it is not a valid number! |
| 18:40.11 | borninbronx | In other words if I re-dial the number a voice answer telling me that it is not valid |
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| 18:45.24 | [TK]D-Fender | ATA's don't have brains to manipulate the numbers they are given |
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| 18:46.06 | [TK]D-Fender | They get what they're given 99% of the time. Any finer points would only be on the rarest of models and only in the most advanced guide they offer. |
| 18:46.23 | [TK]D-Fender | So go check with your manufacturer |
| 18:49.43 | jpsharp | Or your voip provider. |
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| 19:11.31 | borninbronx | [TK]D-Fender: jpsharp the device is a grandstream HT802 and the voip provider is TIM Italy |
| 19:11.45 | [TK]D-Fender | Go read their admin guide |
| 19:12.07 | [TK]D-Fender | but give those don't even support real dilaplans last I checked I'd say you have really low odds of having any options |
| 19:12.38 | borninbronx | [TK]D-Fender: ? what admin guide? you mean the HT802? I've read it but I found no information that could help me.. most parameters do not really explain what they do |
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| 19:13.59 | borninbronx | this is their administration guide: http://www.grandstream.com/sites/default/files/Resources/ht80x_administration_guide.pdf it has some options for caller ID, aside trying all caller ID schemas they have (no difference noticed at all) I've see no option that could help me or I can't understand if it can help |
| 19:14.21 | borninbronx | I was hoping you could give me some resource to understand that manual |
| 19:15.13 | borninbronx | like DTMF Caller ID, I can chose some but I've no idea what that setting does |
| 19:17.38 | [TK]D-Fender | There is no feature to manipulate the number |
| 19:18.03 | borninbronx | So the only thing I can hope for is a firmware update that adds it |
| 19:20.20 | [TK]D-Fender | not happening |
| 19:20.32 | [TK]D-Fender | As I said ATA's aren't supposed to have brains. They do what they're told |
| 19:21.20 | borninbronx | :( does this means I've no way to ditch the ISP provider modem and use my own + a voip device? |
| 19:23.26 | borninbronx | I mean, any ATA device I could buy will have same issue? |
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| 19:24.10 | [TK]D-Fender | <[TK]D-Fender> As I said ATA's aren't supposed to have brains. They do what they're told |
| 19:24.25 | [TK]D-Fender | Your provider shoves a plus sign in there and that's what you'll get |
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| 19:31.30 | borninbronx | tnx [TK]D-Fender kinda sad about this |
| 19:32.39 | [TK]D-Fender | Go use Asterisk then. |
| 19:36.01 | borninbronx | [TK]D-Fender: eheh :) I've free phone calls with my ISP (mandatory with internet connection) and I've had this landline number since forever |
| 19:36.39 | [TK]D-Fender | So far none of that implies yuo can't |
| 19:36.51 | [TK]D-Fender | Do they refuse to give you credentials to connect with your own devices? |
| 19:38.48 | borninbronx | [TK]D-Fender: they've a service to connect using a mobile phone app while you are in the home network.. that app has been reverse engineered and a script is available in python to perform some http calls and obtain new credentials for the voip --- aka: no they don't provide them |
| 19:39.06 | [TK]D-Fender | ~ygwypf |
| 19:39.06 | infobot | i heard ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 19:39.15 | [TK]D-Fender | shrugs |
| 19:39.58 | borninbronx | https://translate.google.com/translate?hl=en&sl=it&tl=en&u=https%3A%2F%2Fwww.ilsoftware.it%2Farticoli.asp%3Ftag%3DVoIP-su-fibra-TIM-con-un-altro-modem-VDSL_14343 |
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| 19:40.58 | borninbronx | it's actually the best internet service available in italy |
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| 23:59.36 | snmp | hi! |