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| 01:36.53 | Kobaz | March 2019 Asterisk User Survey |
| 01:36.53 | Kobaz | Your response has been recorded. |
| 01:36.56 | Kobaz | weeee |
| 01:37.08 | driz | you sure took your time doing that :p |
| 01:37.31 | driz | time sensitive man |
| 01:37.37 | driz | ;) |
| 01:38.21 | Kobaz | It's still March |
| 01:38.23 | Kobaz | I made it... |
| 01:38.31 | driz | :D |
| 01:38.43 | driz | im just teasing man, he only put it up 6days ago |
| 01:38.59 | driz | i wonder if they fixed the issue i reported where you couldn't put git versions |
| 01:39.15 | Kobaz | oh yay |
| 01:39.17 | Kobaz | i found it |
| 01:39.37 | Kobaz | been looking for David Duffett's number for a while, it's actually in my cell phone contacts |
| 01:39.57 | Kobaz | what's the time in the UK |
| 01:40.12 | Kobaz | ew.... okay it's very much AM |
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| 06:21.19 | devdvd | Having a bit of an issue, pretty sure it's nat related. I have an ec2 instance setup with asterisk 16.2.1 using pjsip. I have my client (sip softphone) which is also behind NAT on a different network than asterisk. I have setup a playback extension and I think I can see the problem but can't find how to resolve it. Asterisk is trying to send the RTP data back to the private ip of the client . Here is a pastebin with pjsip.conf, |
| 06:21.36 | devdvd | the console output for the call, the truncated RTP debug output and the pjsip logging output |
| 06:21.40 | devdvd | https://pastebin.com/64HZC2rz |
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| 06:57.12 | bhuddah | devdvd: did you try https://www.voip-info.org/asterisk-sip-nat/ ? |
| 06:57.12 | devdvd | So addition to this: I set my client up through a STUN server and can now get the audio back from the Playback app. However, when I try to use the Echo app, I get nothing and see nothing in the RTP debug |
| 06:57.31 | devdvd | reading |
| 06:57.41 | bhuddah | kinda symmetric nat cheating... |
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| 06:58.15 | devdvd | bhuddah: I'm using pjsip which doesn't seem to have the nat config directive |
| 06:58.52 | bhuddah | any other ways to make your client send out the proper external ip address? |
| 06:59.09 | devdvd | That's what im tyring to figure out |
| 07:00.32 | bhuddah | my understandig is that with stun the client should know its external address. |
| 07:01.13 | devdvd | yeah, that was my understanding as well. That's why im confused about why it works when asterisk sends the audio but not my client |
| 07:02.02 | bhuddah | what stun service are you using? pjnath? |
| 07:02.10 | devdvd | stun.counterpath.com |
| 07:02.21 | bhuddah | oh. not your own stun. okay. |
| 07:02.42 | devdvd | not at this time, however i'm not opposed (and would actually prefer) to setup my own stun |
| 07:03.16 | bhuddah | you could try different alternatives.. like turn or ice... |
| 07:03.32 | bhuddah | and different clients. to see if it's a quirky client. |
| 07:06.19 | realtek | Hello! I install a TDM410p-1p card to server and when i type `dahdi_hardware` command can return the right device. Like this: `pci:0000:04:00.0 wctdm24xxp- d161:8005 Wildcard TDM410P`. And `cat /etc/dahdi/modules` can find `wctdm24xxp`. |
| 07:06.20 | realtek | But when i type `dahdi_cfg` command nothing generate, `dahdi_scan` return nothing. So what problem with this? Any suggestion will be appeciate. |
| 07:06.20 | realtek | dahdi version: 3.0 |
| 07:06.21 | realtek | kernel version: 4.9.0 |
| 07:06.23 | realtek | asterisk version: 16.2 |
| 07:07.35 | devdvd | yeah, using 3 different clients (X-lite, 3cx and zoiper) |
| 07:07.40 | devdvd | all behave the same way |
| 07:08.14 | bhuddah | to me it feels like maybe the stun doesn't do what it's supposed to do. did you already trace it? |
| 07:10.43 | devdvd | possibly, 2 different stun servers have the same behavior though |
| 07:10.50 | devdvd | how would i trace that? |
| 07:12.17 | bhuddah | wireshark... and then a very close look. |
| 07:12.36 | devdvd | ah yeah i've not went that far yet |
| 07:13.08 | bhuddah | please do. all else is just guessing and wasting time. |
| 07:13.38 | devdvd | fair, I wanted to see if there was a config in asterisk i was missing before I went to that. |
| 07:13.46 | devdvd | thanks for the feedback :) |
| 07:14.29 | bhuddah | sure. no prob. i'm just very used to tracing as the very first thing to do. it always wonders me why people don't default to it. |
| 07:15.00 | devdvd | I don't because in my case, it's almost always that i'm missing a config directive somewhere |
| 07:15.12 | drmessano | Stun is almost never necessary |
| 07:15.25 | drmessano | Fix the config |
| 07:15.29 | devdvd | yeah |
| 07:15.35 | devdvd | that's why i asked the question drmessano :) |
| 07:16.08 | devdvd | cuz im not sure what the config should be around this, i've looked through the wiki docs on pjsip with nat and did the things i could find |
| 07:16.22 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT |
| 07:18.06 | drmessano | Lets see the config if you say you've done that |
| 07:19.11 | devdvd | 23:21] <devdvd> https://pastebin.com/64HZC2rz |
| 07:19.31 | devdvd | i know ive missed something, just not sure what |
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| 07:24.01 | drmessano | Are you on the same network as the PBX? |
| 07:24.42 | devdvd | no |
| 07:25.09 | drmessano | Then you can't use the same transport, I believe |
| 07:25.18 | devdvd | the pbx is sitting on an aws ec2 instance and my phone is on my home network behind a nat |
| 07:25.18 | devdvd | ah |
| 07:25.19 | drmessano | Because the signalling and media IPs are not the the same |
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| 07:27.11 | devdvd | hmm, but what would I define as signaling and media for the second connection? Seems kinda strange that i'd have to define my external ip for those as I'd need to do it for every phone that's in a different location |
| 07:27.18 | drmessano | Nothing.. |
| 07:27.24 | devdvd | oh |
| 07:27.49 | devdvd | ok ill give that a shot then |
| 07:28.10 | drmessano | You don't define the media address/signalling address or localnet |
| 07:28.54 | devdvd | for either? |
| 07:29.06 | drmessano | For your clients |
| 07:29.10 | devdvd | ok |
| 07:31.38 | [sID] | Hi all |
| 07:34.39 | [sID] | The question is, how can I issue my carrier 302 to correctly redirect and forward the caller clip? |
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| 07:42.34 | devdvd | drmessano: splitting those out didn't seem to have any effect [phonetrans] type=transport protocol=udp bind=0.0.0.0:5601 then set my transport under the user_defaults section to be phonetrans |
| 07:42.59 | devdvd | same thing as before, asterisk trying to send rtp data back to private ip |
| 07:43.28 | devdvd | i seem to recall having this problem many years ago but of course that was with chan_sip and i can't remember what fixed it :( |
| 07:45.14 | drmessano | Well, my experience with PJSIP is pretty limited⦠WITH CHAN_SIP was a matter of setting the localnet, externhost, and nat=yes. Pretty simple. But none of that applies here |
| 07:45.48 | devdvd | well, im just now getting back into asterisk after a long time and based on the docs i've read pjsip seems to be the new preferred method so I just went with it |
| 07:45.51 | devdvd | is that not true? |
| 07:49.52 | drmessano | chan_sip is dead |
| 07:50.06 | drmessano | No further development, and it's only receiving security fixes from the community |
| 07:55.49 | devdvd | ok |
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| 11:57.47 | vipul20 | Hi, I am using ARI to create an automated call flow. It has multiple playbacks and recordings in the flow. There are some cases where the recordings are completely silent. I am recording the audio in a raw file and it has empty bytes. I did an analysis on over 2400 suspicious recordings and around 340 were like these. I am not sure whether it is because of our PRI or something or is it because of something wrong I did in |
| 11:57.47 | vipul20 | Asterisk. Is there any threshold like thing in Asterisk where certain chunks of audio are being classified as silent? |
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| 12:11.24 | mfalkus | Hi all :) |
| 12:13.22 | mfalkus | I'm wondering if someone can answer an asterisk question for me. In a dialplan, is it possible to get the SIP headers from a REFER message? I know I can get a the refer-to and refer-by headers but I'm after a custom header being sent as well (e.g. X-foo: value). As far as I know this isn't really possible but I'm very much a beginner at asterisk! |
| 12:14.39 | sibiria | with pjsip you should be able to |
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| 12:14.44 | sibiria | using the PJSIP_HEADER function |
| 12:22.44 | mfalkus | Hi sibiria, I think that only gives me back custom headers from the initial invite, rather than the REFER message, but I'll give it another go to double check. |
| 12:24.28 | file | both chan_sip and chan_pjsip only allow initial INVITE header retrieval/manipulation |
| 12:30.05 | mfalkus | ok thanks for the clarification :) |
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| 12:40.42 | sibiria | make uninstall-all does not remove what make config produces |
| 12:41.05 | sibiria | usr-sbin-safe_asterisk and etc-rc.c-init.d/asterisk |
| 12:41.13 | sibiria | rc.d* |
| 12:43.12 | sibiria | the update target of the main makefile does not create the outgoing dir in the spool directory |
| 12:43.17 | sibiria | perhaps this is by choice, not sure |
| 12:44.19 | sibiria | i seem to remember that ast 1.8 created it |
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| 13:16.12 | muAdmDev | I'd like to have my asterisk log files with syntax highlighting in vim. can anyone point me to a possible solution? |
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| 14:39.43 | Helenah1 | Okay so... I have an Asterisk server on my LAN, I have forwarded UDP ports 5060-5061 for SIP, and UDP ports 10000-20000 for RTP because these are the ports used in the Asterisk config. |
| 14:39.57 | Helenah1 | On the LAN, I get Audio, off the LAN I don't get audio. |
| 14:40.40 | Helenah1 | So if I make a call over the internet from one device to another, no audio. we are suspecting it is something to do with RTP, or the NAT, not sure. |
| 14:41.04 | sibiria | sure sounds like NAT. are you using chan_sip or pjsip? |
| 14:41.16 | Helenah1 | pjsip |
| 14:41.55 | Helenah1 | I heard about TURN and STUN relays, I want to avoid these, if I can. |
| 14:42.17 | Helenah1 | It is an extra configuration for myself, and other end users. |
| 14:43.21 | Helenah1 | If it helps, my NAT consists of this: ISP provided NAT ---- DMZ ----- Mikrotik CRS |
| 14:43.52 | Helenah1 | There is no option to set the ISP router to some sort of bridge mode, but I have future plans on replacing it. |
| 14:43.52 | sibiria | i'm not good with pjsip but i have a feeling it has to do with how you define local_net/external_*, along with rtp_symmetric/rewrite_contact settings |
| 14:44.42 | Samot | So the config |
| 14:44.51 | Samot | s/so/show/ |
| 14:44.59 | sibiria | i *think* pjsip will automatically choose between multiple transports (UDP-NAT vs non-NAT), depending on what it sees most fit |
| 14:45.57 | Helenah1 | Okay well, it's my partner working on the Asterisk side of the project, I know more about the network infrastructure. I'll get you a config. |
| 14:46.02 | file | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT |
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| 14:50.00 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:53.06 | Samot | Helenah1: Also for the record, MT's don't really require the port forwarding. |
| 14:53.14 | Samot | They Just Work (TM) |
| 14:54.14 | Samot | As well is this a business ISP connection or a resi/home connection? |
| 14:56.43 | Helenah1 | Samot: http://tinyurl.com/y4wyjwyjpjd |
| 14:56.56 | Helenah1 | Samot: What do you mean by that? |
| 14:57.18 | Samot | That went no where. |
| 14:57.26 | Samot | I mean MTs work awesomely with SIP |
| 14:57.41 | Samot | Unlike most other routers. There really isn't a need to have those port forwards. |
| 14:58.23 | Helenah1 | Samot: http://tinyurl.com/y4wyjwyjpjd/buffer 24 |
| 14:58.28 | Helenah1 | Damn sorry |
| 14:59.02 | Helenah1 | Samot: https://tinyurl.com/y4wyjpjd |
| 14:59.07 | Samot | Just pastebin it. |
| 14:59.13 | Samot | That is not working. |
| 14:59.19 | Helenah1 | I think I'm getting blink, I can't soo the text on my screen very well, my apologies. |
| 14:59.53 | sibiria | should've listened when parents warned about going blind / getting square eyes from all the "computering" |
| 15:00.34 | Helenah1 | Okay... |
| 15:00.57 | Helenah1 | http://paste.ubuntu.com/p/H4dhN2mDRK/ |
| 15:03.25 | Helenah1 | sibiria: I thought RTP ports were a requirement? |
| 15:03.39 | Helenah1 | Er..... Samot |
| 15:05.39 | Samot | No. |
| 15:06.19 | Samot | Which one of these endpoints is the "trunk"? |
| 15:07.12 | sibiria | stateful firewalls don't always keep state for UDP (it's more exception than rule, unless it's an application firewall familiar with SIP/RTP). directing these ports in through the firewall rids you from the uncertainty of not knowing what the firewall does |
| 15:07.27 | Samot | I use MT's 100% of the time. |
| 15:07.36 | Samot | I do not have to had port forwards for SIP signalling or RTP. |
| 15:07.48 | Samot | s/had/add/ |
| 15:08.19 | Helenah1 | All the endpoints are in the dialplan, if that's what you mean. |
| 15:08.32 | Samot | No, I mean which ones of those is your provider endpoint |
| 15:08.43 | Samot | Which one connects you to the outside world for calls... |
| 15:09.03 | Helenah1 | We use GSM, Asterisk is on the LAN, we don't have a SIP provider. |
| 15:09.36 | Samot | OK |
| 15:09.40 | Samot | One last time.. |
| 15:09.47 | Samot | Which one of those endpoints is FOR THAT? |
| 15:09.58 | Helenah1 | None |
| 15:10.24 | Samot | OK |
| 15:10.35 | Samot | So I'm trying to look at your config for this no audio issue with you use the GSM |
| 15:10.41 | Samot | You said you use PJSIP... |
| 15:11.07 | Samot | So what is connecting Asterisk to your GSM? |
| 15:11.21 | Helenah1 | Calls to and from via GSM work, calls to and from within the LAN work, as soon as I do a call over the internet, this is where the audio issue occures. |
| 15:11.39 | Samot | Over the Internet where? |
| 15:11.52 | Samot | Again, what facilitates this? |
| 15:16.43 | pchero_work | Hi, Does 'channel request hangup all' raises HangupRequest event? |
| 15:27.55 | Samot | Alright then. |
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| 15:48.23 | pchero_work | Figured out, it doesn't. It raises SoftHangupRequest event only. |
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| 17:33.55 | driz | any of you know of a SFF pcie fxo card? needs to fit in a 1ru (sideways) |
| 17:36.19 | driz | closest i've found is an atcom low profile card, but much like most of them, they're prefabbed to support 4 fxX which makes the card excessively long |
| 17:36.40 | driz | seems like everyone is using the same long ass base board |
| 17:41.41 | Samot | Nope. I don't use internal cards. |
| 17:42.11 | driz | you have a reliable usb adapter you use? |
| 17:42.27 | Samot | I use gateways. |
| 17:42.30 | Samot | No USB |
| 17:42.32 | Samot | Ethernet. |
| 17:42.39 | Samot | SIP. |
| 17:43.14 | driz | yah i dont want my security system on SIP, bad practice |
| 17:43.26 | driz | and my VG310 is full :/ |
| 17:43.29 | driz | oh well |
| 17:43.37 | Samot | What? |
| 17:43.47 | Samot | How is it a bad practice? |
| 17:44.37 | driz | well, ill tldr for you.. what happens when your WAN goes down to you SIP session? or power goes out, ups dies, generator runs out of fuel |
| 17:45.02 | driz | it's pretty well accepted by the industry that gas meters, red phones, and security lines belong on 1FBs typically speaking |
| 17:45.46 | driz | but that's besides the point; im merely looking for a card that fits, not your 'ethernet. Sip.' |
| 17:46.05 | Samot | I answered your question. |
| 17:46.11 | Samot | Then you asked another, I answered that. |
| 17:46.13 | Samot | That's it. |
| 17:46.16 | driz | you literally did not |
| 17:46.35 | driz | i mean ok, you answered like a 5th grader "no." |
| 17:46.35 | driz | lol |
| 17:46.38 | Samot | any of you know of a SFF pcie fxo card? <-- Nope, I don't use internal cards. |
| 17:46.40 | driz | im just gonna block you man |
| 17:46.43 | driz | take care |
| 17:46.50 | Samot | 1:42:13 PM <driz> you have a reliable usb adapter you use? |
| 17:46.51 | Samot | 1:42:28 PM <Samot> I use gateways. |
| 17:46.55 | Samot | OK |
| 17:46.58 | Samot | Have a good day. |
| 17:47.32 | driz | Anyway, if anyone knows of a SFF pcie fxo card please let me know! thanks |
| 17:49.02 | Kobaz | driz: yeah please define 'bad practice' with using sip and X |
| 17:49.13 | driz | literally just did |
| 17:49.36 | sibiria | driz' perspective is not complicated: he want something close to a dedicated line that doesn't depend on a bunch of external factors |
| 17:49.37 | Samot | Not really but anyways.. |
| 17:49.45 | driz | correct |
| 17:50.00 | Samot | sibiria: There is a difference between a bad practice and a poor setup. |
| 17:50.09 | Samot | So let's not equate them. |
| 17:50.22 | Kobaz | 'bunch of external factors' like, having power to the unit? |
| 17:50.22 | Samot | Also, I don't think anything less than a 4 port card exists anymore. |
| 17:50.24 | Samot | Because why? |
| 17:50.34 | driz | or the many items i literally just typed |
| 17:50.35 | sibiria | Kobaz: like a bunch of interconnecting network operators |
| 17:50.40 | sibiria | ie. basic WAN route |
| 17:50.46 | Samot | I never said a WAN route. |
| 17:50.50 | Kobaz | sibiria: and PCI is basically an interconnected network |
| 17:50.55 | Samot | I said a gateway |
| 17:50.58 | Kobaz | you're trading one interconnected network with a nother |
| 17:51.05 | Samot | An FXO to SIP Gateway that connects SIP to the PBX. |
| 17:51.05 | driz | trading one i control |
| 17:51.11 | driz | for one various ISPs control |
| 17:51.22 | Samot | Since the PBX will still need power if there is an outage, there will be a UPS/backup. |
| 17:51.23 | driz | it's pretty simple dude |
| 17:51.34 | Kobaz | you can control your network if you like. if you want someone else to control it, that's your choice |
| 17:51.40 | Samot | I don't even know how a WAN connection got involved. |
| 17:51.42 | driz | you control your own MPLS? |
| 17:51.47 | driz | i think you're a liar |
| 17:51.52 | Kobaz | just saying don't limit your options based on a misunderstanding of the technology |
| 17:52.00 | driz | I'm not misunderstanding anything |
| 17:52.18 | Kobaz | why would you need a WAN connection to have a local appliance talk to a local sip gateway? |
| 17:52.18 | driz | this is standard best practice whether you work on nortel, avaya, or cisco |
| 17:52.33 | driz | dude, im not even talking about your hobby tier stuff |
| 17:52.35 | driz | just forget it |
| 17:52.36 | driz | lol |
| 17:52.52 | Kobaz | and nortel and avaya both filed for bankruptsey, so your point being? |
| 17:53.05 | driz | lots of ppl file for bankruptcy, what's your point? |
| 17:53.12 | driz | chapter 11 reorg isn't a big deal |
| 17:53.23 | driz | it's common business practice and makes sense frequently from a business perspective |
| 17:53.37 | Samot | Kobaz: Forget about it. MPLS has just be equated as a service from them... |
| 17:53.37 | driz | but since you focused on those, cisco hasn't filed bankruptcy |
| 17:53.49 | Kobaz | sometimes it is, sometims it isn't, but 'just because XYZ do it this way'. is not necessarily a reason |
| 17:53.50 | Samot | Kobaz: No need for your hobbyist stuff.. |
| 17:53.57 | driz | i listed the reasons |
| 17:54.06 | driz | you just seem to think your sip calls work without wan |
| 17:54.09 | Kobaz | Samot: not sure what you're talking about, we run thousands of phones |
| 17:54.13 | Samot | I never got a rebuttal on those reasons either, he blocked me. |
| 17:54.18 | driz | and maybe they do, when you call phones on your LAN |
| 17:54.32 | Samot | Kobaz: MPLS is not a service. It's a technique. |
| 17:54.42 | Kobaz | sure sure |
| 17:54.46 | Samot | Kobaz: Like how Ring Groups are a real thing, it's a method. |
| 17:55.00 | Kobaz | right... and i wasn't talking about any of that? |
| 17:55.15 | Samot | No, he slammed you for being a hobbyist. |
| 17:55.22 | Samot | I'm kinda defending you... |
| 17:55.32 | Kobaz | driz: my sip calls work fine without a WAN, depending on what is servicing those sip calls, and where they are coming from and going to |
| 17:55.32 | driz | to be fair, this guy just said you don't need WAN for phones to work |
| 17:55.37 | Kobaz | Samot: oh okay |
| 17:55.38 | driz | that's hobbyist tier |
| 17:55.42 | driz | im calling OUTSIDE of my LAN |
| 17:55.47 | sibiria | i think a key here to understanding driz' perspective is that he - unless i misunderstand - prefers to do these calls over the POTS? |
| 17:55.57 | driz | only security systems and red phones |
| 17:55.59 | Kobaz | maybe he does want a pots line? |
| 17:56.00 | sibiria | which really is a pretty damned reliable network |
| 17:56.12 | driz | and gas meters of course |
| 17:56.16 | Samot | I think he misunderstood what I was taking about... |
| 17:56.26 | Kobaz | Samot: correct |
| 17:56.36 | Samot | I was referring to an FXO gateway to take the POTS to SIP internally |
| 17:56.41 | Samot | Between the Gateway and the PBX |
| 17:56.46 | Samot | Nothing about doing SIP trunks. |
| 17:57.03 | Kobaz | Right, that's fine and dandy |
| 17:57.07 | Kobaz | and you control all of it |
| 17:57.11 | Samot | That's what I suggested.. |
| 17:57.17 | Samot | And that's what started this rant of his. |
| 17:57.18 | Kobaz | you can do a direct ethernet connection to avoid any switches |
| 17:57.21 | Kobaz | right |
| 17:57.22 | Samot | Right. |
| 17:57.23 | driz | i didnt rant |
| 17:57.25 | driz | i ignored you |
| 17:57.25 | sibiria | right |
| 17:57.25 | Samot | All local. |
| 17:57.29 | Kobaz | he doesn't like ethernet is the issue maybe? |
| 17:57.32 | Samot | I never said WAN |
| 17:57.37 | driz | and kobaz chimed in with an arguement from the hobbyist perspective |
| 17:57.40 | Samot | I never said anything about outside the network. |
| 17:57.43 | Kobaz | he wants to connect via PCI instead of ethernet, |
| 17:57.52 | Kobaz | driz: i have no idea what you're talking about with this hobbiest perspective |
| 17:57.57 | sibiria | you guys let me know when we're ready to group-hug this thing out |
| 17:58.10 | driz | im good to hug now |
| 17:58.16 | driz | it's cold here |
| 17:58.24 | sibiria | \o/ |
| 17:58.32 | driz | \o/ |
| 17:58.43 | driz | 60F is too cold for server rooms :( |
| 17:59.04 | Kobaz | Just saying a 4 port gateway is just as reliable as an internal pci card |
| 17:59.15 | Samot | Yes, I agree. |
| 17:59.17 | *** join/#asterisk eharris (~eharris@unaffiliated/eharris) |
| 17:59.19 | Kobaz | having run both for more than 15 years and having an equal failure rate with either |
| 17:59.36 | Samot | And you're not running around looking for a card that fits your system. |
| 17:59.40 | Kobaz | right |
| 17:59.41 | driz | i'd have to see the MTBF on the gateway you're suggesting and compare it to the mtbf of a pci card |
| 17:59.52 | driz | i work at the enterprise level, MTBF is everything |
| 17:59.55 | Samot | I'm not suggest a particular one. |
| 17:59.59 | Kobaz | and if something DOES happen with the gateway/port, you replace the gateway and don't have to open up the box and shut down the system |
| 18:00.02 | Samot | I was suggesting it as an alternative. |
| 18:00.10 | Samot | Which you can look for the right kinda. |
| 18:00.16 | driz | ANY 4port gateway with an MTBF over 500k hours, i think would be sufficient |
| 18:01.05 | Kobaz | a lot of those specs aren't widely published, sadly |
| 18:01.11 | driz | i think you'll be hard pressed to find an external device (the cisco vg350 does have an MTBF over 1mil hours, but it's quite a lot of ports) |
| 18:01.16 | driz | with an MTBF that high |
| 18:01.17 | Samot | I've been using the same gateway at places for years. |
| 18:01.42 | Samot | I rarely have to replace them. |
| 18:01.50 | Kobaz | driz: all those MTFBs are calculated anyway |
| 18:02.23 | driz | most things are calculated dude :p |
| 18:02.30 | driz | it's why it's a MEAN time |
| 18:02.45 | driz | it's the mean of all reported or discovered failures |
| 18:03.12 | Kobaz | right |
| 18:03.28 | Samot | Again, I have gateways that have been in place for years.. |
| 18:03.30 | Samot | No issues. |
| 18:04.26 | Samot | I mean 500K hours is 57 years. |
| 18:04.39 | Kobaz | right |
| 18:04.48 | Kobaz | and you're going to be replacing equipment in 10-15 anyway |
| 18:05.01 | Samot | So you're looking of a device with a 57+ year MTBF? |
| 18:05.27 | Kobaz | sangoma line cards do have a 1m MTBF for PRI and such |
| 18:05.37 | driz | you'd hope you replace in 10-15 i've seen some frighteningly old PBXs that ppl use in production still; |
| 18:05.44 | driz | i have found that the PRI card MTBF is quite high |
| 18:05.51 | Kobaz | oh sure, we have a customer with 6000 lines on a Definity version 4 |
| 18:06.43 | Samot | I still have mitels from the 70's in place. |
| 18:07.02 | Kobaz | It's gotta be like 20 years old |
| 18:07.07 | driz | i just ripped out a mitel |
| 18:07.09 | driz | and |
| 18:07.09 | Kobaz | yeah, some of that stuff just chugs away |
| 18:07.13 | driz | replaced with a shoretel :( |
| 18:07.18 | Kobaz | meh |
| 18:07.24 | driz | i felt like something went wrong in the client's decision process |
| 18:07.26 | *** join/#asterisk t1k3 (~t1k3@pool-71-112-160-141.pitbpa.fios.verizon.net) |
| 18:07.48 | driz | tbf the shoretel was maybe 7 yrs old and the mitel was 25yrs old |
| 18:07.53 | driz | but shit. |
| 18:08.43 | driz | 90% of my work is asterisk or cisco call manager; stresses me out when i have to touch nortels, mitels, or avayas |
| 18:09.53 | Samot | So you need a single FXO port for this PBX? |
| 18:10.29 | Kobaz | and it needs to be SFF ? |
| 18:10.41 | driz | preferably, they're using 1ru supermicros |
| 18:10.43 | driz | it's a tight fit |
| 18:10.54 | Kobaz | good luck |
| 18:11.00 | Samot | But just a single FXO port? |
| 18:11.07 | Samot | That's going to be your issue right there. |
| 18:11.13 | driz | that's what im finding |
| 18:11.21 | Samot | They basically don't make anything less than 4 now. |
| 18:11.21 | driz | every base board is intended for 4 ports |
| 18:11.34 | Samot | Be it a FXO/FXS combo or straight up on or the other. |
| 18:11.43 | driz | i found a pci card with 1port, but the adapter to get pcie makes the card too long |
| 18:11.46 | driz | yah |
| 18:11.53 | Kobaz | you might be able to find one of those old clones |
| 18:11.59 | Kobaz | it's basically a modem with some extras |
| 18:12.07 | driz | yah, those x100 knockoffs |
| 18:12.09 | Kobaz | the signal quality is pretty bad |
| 18:12.36 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
| 18:12.46 | Kobaz | https://www.aliexpress.com/item/X100P-1Port-FXO-card-with-1-FXO-module-TDM400P-TDM410P-TDM400-A400P-AX400P-X100-Analog-Card/695344453.html |
| 18:13.06 | driz | this runs into my issue |
| 18:13.06 | Samot | So you need to do this for multiple 1RU micro's? |
| 18:13.09 | Kobaz | and there goes your 1 million MTBF out the door |
| 18:13.12 | driz | to use that, i need a pcie adapter |
| 18:13.16 | Kobaz | some random made in china thing |
| 18:13.20 | driz | which further adds single point of failure |
| 18:13.23 | driz | and reduces mtbf more |
| 18:13.29 | driz | plus the size issue with the adapter |
| 18:13.34 | driz | i HATE 1ru servers |
| 18:14.04 | Samot | What are these servers for? Just need a POTS line for emergency? |
| 18:14.35 | driz | some water heater company; it's for their security system |
| 18:14.50 | Samot | So yeah, gateway. |
| 18:15.24 | driz | yah, i have an ata192 on the site now |
| 18:15.32 | driz | they just suck |
| 18:15.37 | driz | oh well |
| 18:15.41 | driz | is what it is i suppose |
| 18:15.51 | driz | i told them to just plug the security system directly in to the pots |
| 18:16.10 | driz | but the security company told them that wasn't necessary |
| 18:16.27 | driz | well anyway |
| 18:16.33 | driz | i enjoyed the arguments guys |
| 18:16.37 | driz | kobaz |
| 18:16.41 | driz | you're a filthy hobbyist |
| 18:16.47 | Kobaz | fun |
| 18:16.47 | driz | \o/ but i hug you anyway |
| 18:16.49 | driz | :D |
| 18:16.59 | Samot | Wait... |
| 18:17.00 | driz | at least i didn't call you a user dude |
| 18:17.01 | Samot | 192's? |
| 18:17.05 | driz | yah |
| 18:17.40 | Kobaz | those guys are cute |
| 18:17.51 | Kobaz | i put one at my parents house |
| 18:17.52 | Samot | Those are FXS. |
| 18:18.00 | Samot | You started this by asking about FXO. |
| 18:18.04 | Samot | Not the same. |
| 18:18.08 | Kobaz | haha |
| 18:18.10 | Kobaz | good point |
| 18:18.13 | driz | oh i know |
| 18:18.17 | driz | the 192 doesnt work |
| 18:18.36 | driz | they had them hooked up to their valcom paging units |
| 18:18.40 | driz | and decided it would work the same |
| 18:18.51 | driz | "it's the same type of phone line" |
| 18:19.18 | Kobaz | your situation sounds all kinds of messed up |
| 18:19.26 | driz | my client is all kinds of messed up |
| 18:19.35 | driz | they need to just plug the system into the wall |
| 18:19.41 | driz | not run the wall into the pbx into the system |
| 18:19.55 | driz | but they just argue about everything |
| 18:20.05 | driz | "reddit says do this, so this is what we want" |
| 18:20.10 | Kobaz | soooo, they have a fxs coming in from the carrier, and they don't want to use that? |
| 18:20.10 | driz | ^ actual quote. |
| 18:20.15 | driz | yes |
| 18:20.16 | Kobaz | that sounds like the ideal solution |
| 18:20.16 | Samot | I use Cisco SPA112's for that stuff. |
| 18:20.22 | driz | they purchased a 1FB |
| 18:23.07 | driz | i like a lot of cisco's small business stuff, but the client (a small business) refuses to use "that linksys shit" regardless of how many links i give them showing not every cisco small business item was made by linksys |
| 18:23.36 | driz | honestly, once this project is over, im done with this client |
| 18:27.25 | driz | i gotta run to a meeting |
| 18:27.28 | driz | cya guys |
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| 19:34.38 | Silicium | hi all, i'am about to start traveling around the world, starting in europe and frist stop in southeast asia. I'am lookig for Aid agencies in and around Asia where i can help mit my IT, Network Infrastructure and Electricity knowledge. if you know such organisations, please let me know! cheers |
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| 20:01.43 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 20:13.00 | rpifan | Silicium, what |
| 20:13.40 | *** join/#asterisk toerkeium (be0237fa@gateway/web/freenode/ip.190.2.55.250) |
| 20:16.41 | toerkeium | [TK]D-Fender: hello. If you remember me from yesterday, for the issue of "sometimes no audio" when calling, I believe it was a sum of things. My trunk, was setup to deny=all and allow=g729, allowing more codecs, some specific calls started to work without any issue. Also, we have a Mikrotik router, which does have a SIP helper service. Disabling this in the router, improved performance |
| 20:17.39 | [TK]D-Fender | codec negotiation failure will kill a call instantly, not cause "no audio" and let the call continue |
| 20:18.26 | toerkeium | [TK]D-Fender: can't tell for sure but, applying those "fixes" looks it's working well now.. so far |
| 20:18.34 | toerkeium | just wanted to let you know |
| 20:19.12 | toerkeium | also added same LAN IP to the pbx as the workstations |
| 20:19.22 | toerkeium | shouldnt make any change but, to be sure.. |
| 20:19.27 | [TK]D-Fender | huh? |
| 20:19.52 | toerkeium | pbx wasn't in the same LAN as the workstations |
| 20:20.03 | toerkeium | we use softphones |
| 20:20.10 | toerkeium | workstatinos == softphones |
| 20:21.02 | *** join/#asterisk rpifan__ (~rpifan@45.77.199.32) |
| 20:22.58 | toerkeium | the directmedia setting makes voip phones audio point to point? |
| 20:23.18 | toerkeium | it bypass the pbx completely? |
| 20:24.12 | [TK]D-Fender | if you allow both sides to do it, yes |
| 20:24.22 | [TK]D-Fender | if you span across NAT then thing start going wrong |
| 20:24.36 | [TK]D-Fender | So generally better to just force it all through the server |
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| 20:25.23 | toerkeium | great, thanks |
| 20:31.20 | Silicium | rpifan__: what i said |
| 20:41.58 | rpifan__ | Silicium, and why |
| 20:44.41 | Silicium | because its good to help |
| 21:02.53 | *** join/#asterisk anexit (~anexit@push.anexit.net) |
| 21:03.08 | anexit | Question, should voip phones use UDP or TCP and why? |
| 21:04.18 | rpifan__ | udp |
| 21:04.26 | rpifan__ | u r german |
| 21:04.46 | seanbright | well it depends on the protocol, but assuming sip it uses both |
| 21:04.56 | seanbright | it's supposed to, anyway |
| 21:05.50 | anexit | Wouldn't tcp be the better protocol for sip as it has packet inspection and cross checking? |
| 21:07.03 | seanbright | what is cross checking? |
| 21:07.19 | seanbright | guaranteed ordering? |
| 21:07.26 | anexit | Datagrams are in order and guranteed |
| 21:07.27 | anexit | yea |
| 21:08.56 | anexit | I was just wondering, I know udp is fast as it only is about 8bytes a packet and TCP runs about 120bytes. |
| 21:10.10 | seanbright | i don't know the rationale for using UDP for SIP. i have some assumptions, but nothing definitive. |
| 21:11.35 | seanbright | from the original SIP RFC: UDP allows the application to more carefully control the timing of messages and their retransmission, to perform parallel searches without requiring TCP connection state for each outstanding request, and to use multicast. Routers can more readily snoop SIP UDP packets. |
| 21:12.04 | seanbright | (obsolete RFC2543) |
| 21:16.16 | seanbright | SIP is only chatty at the beginning and end of a session, so it's wasteful to keep a TCP connection around |
| 21:17.06 | anexit | yeah |
| 21:17.14 | sibiria | wasteful in one way, but keeping a state open isn't that costly |
| 21:17.41 | sibiria | (it may obviously be if you run a very large operation on a single host) |
| 21:17.43 | anexit | Not really wasteful as with stateful it can reuse itself |
| 21:18.55 | seanbright | you can use SIP over TCP if you like |
| 21:19.25 | anexit | yeah I migth try it. |
| 21:20.06 | seanbright | when you say "can reuse itself" what do you mean exactly? |
| 21:20.26 | seanbright | multiple sessions over a single TCP connection? |
| 21:21.13 | seanbright | i don't think asterisk does that, but i haven't really looked into it |
| 21:23.48 | anexit | Just thinking from a router point of view, when a session over tcp is established it hangs around for a bit but it keeps it open incase that transmission is inbound/outbound again. |
| 21:24.23 | anexit | So initiation would be fast if the caller makes that call to the same user. |
| 21:25.03 | seanbright | again, i don't know that aserisk will re-use a TCP transport connection for multiple calls |
| 21:25.08 | sibiria | would be fast either way, really :) |
| 21:25.22 | seanbright | frankly, i don't think you gain anything from using TCP over UDP |
| 21:25.40 | anexit | well.. udp still needs to create a state so yeah |
| 21:25.44 | sibiria | the most obvious problem with using TCP for SIP is that plenty of soft-phones only do UDP |
| 21:25.53 | sibiria | because it's the de facto standard |
| 21:26.30 | seanbright | then they aren't rfc compliant |
| 21:26.36 | anexit | But that could be classified in a seperate phone network. |
| 21:26.49 | anexit | If it is the case |
| 21:27.43 | sibiria | seanbright: yep, that is correct |
| 21:28.51 | drmessano | Is it REALLY "plenty of soft-phones" at this point? With the adoption of TLS, many soft-phones support TCP/TLS |
| 21:28.58 | drmessano | Been that way for a while now |
| 21:29.13 | drmessano | I can't say the last time I ran across one that was UDP only |
| 21:30.17 | sibiria | it wasn't that long ago Linphone was UDP only when using unencrypted transport |
| 21:30.27 | sibiria | (just to mention a very popular soft-phone) |
| 21:31.07 | sibiria | Telephone (macOS) is UDP only |
| 21:32.01 | drmessano | IDK, I see bug reports for TCP issues with Linphone from 2014 |
| 21:32.08 | drmessano | So that's at least 5 years |
| 21:32.40 | seanbright | or you could just use UDP |
| 21:32.48 | seanbright | like a normal person |
| 21:34.46 | anexit | Ah yeah, TLS is a must these days |
| 21:35.16 | anexit | Bria is decent and yate |
| 21:36.01 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 21:36.14 | seanbright | eh |
| 21:36.22 | seanbright | good times |
| 21:36.28 | drmessano | sibiria: Also, telephone does support TCP |
| 21:36.48 | drmessano | Honestly not having TCP/TLS support is an old problem |
| 21:37.53 | sibiria | Telephone supports TCP if you proxy |
| 21:37.56 | sibiria | not otherwise |
| 21:39.57 | seanbright | now he's off trying to prove you wrong |
| 21:40.03 | seanbright | this is what the internet is now |
| 21:40.06 | drmessano | No, I know he's wrong |
| 21:40.07 | sibiria | it supports TLS, but i'm talking about unencrypted transport |
| 21:40.11 | drmessano | No point arguing |
| 21:41.13 | drmessano | I'm also talking about unencrypted transport, else I would have specified "TLS" |
| 21:42.00 | drmessano | But anyway.. Lack of TCP support is an old problem. Haven't run across a popular, maintained client that's lack it in years |
| 21:42.02 | sibiria | Telephone can do TCP when you configure a proxy, otherwise it will enforce UDP and consequently time out after failure to initiate. the end. |
| 21:42.47 | sibiria | well, one could argue Telephone is not a "Top Dog" among the soft-phones, sure. it's definitely one of those unseen applications |
| 21:43.21 | drmessano | I wasn't making that argument |
| 21:43.33 | sibiria | ok but if you would like to, i just laid it out for you |
| 21:43.34 | sibiria | :) |
| 21:43.54 | drmessano | Because had I left off that part, you would have found some 10 year old unmaintained client without TCP and said "AH HA!" |
| 21:44.06 | sibiria | Telephone is not unmaintained |
| 21:44.12 | drmessano | Nor did I say it was |
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| 22:51.08 | jpsharp | It's just a bit naggy. |
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