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| 00:36.02 | darkdrgn2k | hi all, anyone know of any decent european sip providers. Looking for incoming calls specifically. |
| 00:40.56 | sibiria | infobip is ok |
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| 04:13.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.4.0, 16.18.0 (2021/05/06) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 09:22.48 | UncleKiwi | Hola |
| 09:23.33 | UncleKiwi | for some reason my dialplan is not showing when I type dialplan show - im using asterisk v18 |
| 09:47.29 | UncleKiwi | ok solved - a file from my dial plan was missing |
| 10:02.22 | UncleKiwi | oh no - i tried to run voip over a wireless ISPs network connection and its giving audio that is choppy |
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| 10:13.44 | UncleKiwi | i have just tried adding jbenable=yes jbforce=yes and it sounds like it has helped a little - is there any settings that i can try to help with what i assume to be a jittery conection |
| 10:29.12 | UncleKiwi | is choppy audio just bad news that is probably the internet connection and there is little i can do about it ? |
| 10:29.29 | UncleKiwi | i have seen this situation before with this ISP |
| 10:30.10 | UncleKiwi | maybe the packets arrive out of order or slow or something - its just a mess |
| 10:31.08 | UncleKiwi | pings report about a 50ms to the voip provider |
| 10:31.21 | UncleKiwi | normally 50ms is not an issue |
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| 10:38.28 | orn | UncleKiwi: Likely just a jitter that is too big to solve. You could try to increase the size of the jitter buffer (and maybe look at jitter buffer settings on the client side as well), but generally speaking the advice is to deal with the actual source of the problem |
| 10:41.24 | UncleKiwi | orn: looks quite tricky to configure - |
| 10:41.45 | UncleKiwi | do you mean like i should replace the internet connection |
| 10:41.50 | UncleKiwi | or talk to the ISP |
| 10:42.08 | UncleKiwi | something like that |
| 10:43.22 | orn | UncleKiwi: Are there no wired options available at all at this location? |
| 10:44.05 | orn | UncleKiwi: Also, what type of wireless ISP connection is this? WiFi? 4G? Maybe look at placing the antenna at a better location. If a simple 4G router, maybe get one with an outdoor antenna. |
| 10:44.37 | UncleKiwi | orn: yeah there is but its low bandwidth - we have wireless connection there which is fine for maybe other stuff and delivers good throughput - I think i will have to install adsl2 |
| 10:44.40 | orn | but yeah, something along those lines -- deal with the underlying problem by any means possible before trying to solve it with a jitter buffer |
| 10:46.15 | orn | FWIW, I work for a mobile focused operator and we generally don't sell VoIP services over 4G/5G unless the customer flat out refuses other solutions |
| 10:47.13 | UncleKiwi | its not over 4g its over some 5Ghz wireless maybe ubiquity |
| 10:47.17 | UncleKiwi | hardware |
| 10:48.23 | orn | UncleKiwi: In my opinion, that is way worse, since that is an open spectrum liable to all sorts of interference |
| 10:48.40 | orn | is it a point-to-point connection with a directional antenna? |
| 10:49.28 | UncleKiwi | it would be point to multipoint - it will be a sector on a hill with abackhaul |
| 10:49.45 | UncleKiwi | anyway yeah... its a fail for voip unfortunatly |
| 10:50.10 | orn | UncleKiwi: I see. It's very difficult to guarantee any sort of response rates on such connections. You might get all sorts of spikes. |
| 10:50.22 | orn | the best you can hope for is an adaptive jitter buffer with a high maximum setting |
| 10:50.44 | orn | you could just run ping for a while to see how high it spikes, and maybe use that as your max value |
| 10:50.56 | orn | but you might be looking at quite the delay |
| 10:52.06 | UncleKiwi | :) it working |
| 10:52.13 | UncleKiwi | Set(JITTERBUFFER(adaptive)=700,,60) |
| 10:54.37 | orn | all righty -- if that works for you, great :) user's might experience some half-duplex radio type conversations though :) |
| 10:54.52 | orn | "hello, are you there, OVER" :) |
| 10:56.02 | UncleKiwi | im testing it again and again by calling the number and having ti play the tt-monty-knights |
| 10:59.30 | UncleKiwi | umm... yeah its not fixed |
| 11:00.04 | UncleKiwi | its just wireless and its moments of good conditions ie lack of congestion or interference |
| 11:09.36 | file | jitterbuffer only covers media from a channel, it doesn't help if it's media sent to a channel |
| 11:09.43 | file | fyi |
| 11:10.06 | file | in that case it is the responsibility of the receivers jitterbuffer to do things |
| 11:12.43 | UncleKiwi | i have two asterisk boxes one is connected to a better internet connection - maybe I could use IAX between them and take advantage of the jitterbuffer ? |
| 11:14.19 | UncleKiwi | maybe its just better i dont use this wireless connection for voip |
| 11:24.58 | orn | UncleKiwi: that would be optimal. i.e. not using the wireless connection. but as mentioned by file both sides need to use a jitter buffer |
| 11:25.53 | file | There are also situations where even a jitterbuffer can't help, as the buffering required just makes the experience poor |
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| 11:27.05 | UncleKiwi | thanks for this info. |
| 11:27.32 | UncleKiwi | it seems to me that a quiality internet connection is key |
| 11:28.37 | UncleKiwi | the tt-monty-knights is quite an abusive audio - I have had a lot of farts in my general direction |
| 11:33.47 | orn | hahaha |
| 11:33.50 | orn | indeed |
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| 12:16.14 | Samot | Ive run voip over WISPs |
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| 12:33.14 | UncleKiwi | Hi Samot, did you have any trouble ? |
| 12:33.41 | Samot | In general, no. Have I, yes. The quality of the WISP is always a factor. |
| 12:33.44 | UncleKiwi | it seems to work its just that the audio is choppy |
| 12:33.58 | Samot | What speeds does this provide? |
| 12:34.13 | UncleKiwi | 30Mbit down 5Mbit up |
| 12:34.28 | UncleKiwi | 50ms latency |
| 12:34.45 | UncleKiwi | to the voip server |
| 12:35.17 | Samot | And every call is choppy? |
| 12:35.25 | UncleKiwi | nah not all calls |
| 12:35.40 | Samot | So how many calls? |
| 12:36.01 | UncleKiwi | i am assuming its network congestion on the wireless network (ISPs network) |
| 12:36.28 | UncleKiwi | i have just beeing calling the number and getting a test audio played |
| 12:36.34 | UncleKiwi | i have probably called it maybe 50 times |
| 12:37.04 | UncleKiwi | i just called it now and it was perfect |
| 12:37.16 | Samot | And out of those 50 times how many where choppy audio? |
| 12:37.37 | UncleKiwi | maybe 33% |
| 12:37.58 | UncleKiwi | maybe 50% |
| 12:38.03 | UncleKiwi | its better now |
| 12:38.20 | UncleKiwi | time here is amost 1am |
| 12:38.33 | UncleKiwi | so network is probably not in use |
| 12:38.46 | Samot | Is this a WISP network or mobile? |
| 12:38.56 | UncleKiwi | WISP |
| 12:39.46 | UncleKiwi | i think thery are using 5g ubiquity air max |
| 12:40.40 | UncleKiwi | i suspect its network congestion on the sector |
| 12:41.03 | UncleKiwi | because as i test it now at 1am its perfect |
| 12:41.25 | UncleKiwi | 12:41 |
| 12:42.18 | Samot | OK so this really doesn't have to do with being wireless. |
| 12:42.26 | Samot | This has to do with the ISP being oversold. |
| 12:42.38 | Samot | Easy to blame wireless. |
| 12:42.49 | UncleKiwi | probably - aye |
| 12:43.39 | UncleKiwi | i mean ' |
| 12:43.46 | Samot | Well during "busy hours" you have choppy audio and probably other issues. Off hours, it's fine. |
| 12:43.49 | UncleKiwi | 'normal' traffic at this time |
| 12:44.04 | UncleKiwi | it not normal traffic at this hour |
| 12:44.17 | UncleKiwi | yeah |
| 12:44.39 | UncleKiwi | im going to heat up a doughnut - just been to the supermarket |
| 12:51.12 | UncleKiwi | i have a really fat customer and he twists up the telephone cord and destroys it |
| 12:51.24 | UncleKiwi | about every 6 months |
| 12:59.48 | UncleKiwi | when using IAX2 to connect two asterisk PBXs by default its not encrypted - passwords would not be sent over the net in plain text right ? |
| 13:01.15 | Samot | Shrug. I don't use IAX because, why. |
| 13:01.40 | UncleKiwi | IAX is useful |
| 13:02.23 | UncleKiwi | for connecting a couple offices together |
| 13:02.42 | UncleKiwi | thats what i use it for |
| 13:03.14 | UncleKiwi | internal calls and BLF can be passed thru the IAX |
| 13:12.27 | Samot | Same with SIP |
| 13:16.00 | UncleKiwi | yeah i think SIP can do it also |
| 13:17.05 | UncleKiwi | but IAX sends all the sigaling and audio all over one udp port - so rtp etc is not a concern |
| 13:17.14 | Samot | No SIP can do it |
| 13:17.28 | Samot | Ok. Those are 2005 issues. |
| 13:18.00 | orn | meh... I still see a fair share of such issues -- people have old CPEs all over the place |
| 13:18.17 | orn | and even with newer ones (fortigate comes to mind) |
| 13:18.26 | Samot | Ok |
| 13:18.41 | orn | but there's other ways to remedy those, such as VPN or Wireguard etc |
| 13:18.50 | orn | UncleKiwi: you might want to look at a wireguard link between the pbxs |
| 13:18.56 | Samot | So if this was such a huge problem why is IAX an Asterisk only option? |
| 13:19.18 | orn | who said it was the only option? |
| 13:19.31 | Samot | I didnt |
| 13:19.44 | Samot | I asked why have more adopted it? |
| 13:19.49 | Samot | Havent |
| 13:20.15 | orn | did you? |
| 13:20.21 | orn | i didn't see that |
| 13:20.32 | Samot | I asked why it was still an Asterisk only thing |
| 13:20.57 | Samot | If SIP was so problematic and IAX solves said problems..why is SIP king? |
| 13:21.45 | UncleKiwi | IAX is used to join the two asteriusk boxes - it can transport sip over the IAX link |
| 13:21.55 | UncleKiwi | well sip calls |
| 13:22.02 | UncleKiwi | thats what I do |
| 13:22.15 | UncleKiwi | IAX has its place and so does SIP |
| 13:22.26 | UncleKiwi | IAX is great for asterisk to asterisk |
| 13:22.35 | Samot | So is SIP |
| 13:22.46 | UncleKiwi | ok so use SIP |
| 13:22.56 | Samot | I do. |
| 13:22.59 | UncleKiwi | I find IAX more simple |
| 13:23.18 | Samot | Because IAX doesnt solve any problems for me |
| 13:23.21 | Samot | Never has. |
| 13:23.22 | UncleKiwi | in the asterisk to asterisk connections |
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| 13:25.08 | orn | Samot: Ah, now I understand what you mean. I misread your sentence. I didn't take it to mean "Why isn't IAX available outside of Asterisk" |
| 13:26.15 | orn | SIP was first is the simplest answer, and IAX is proprietary is the second answer. A compounding factor of those two. Also, IAX brings other problems to the table and doesn't really solve that many. |
| 13:26.28 | Samot | Right. |
| 13:26.37 | Samot | And IAX had an RFC draft |
| 13:26.49 | Samot | It was in the protocol war |
| 13:27.34 | orn | Yeah -- the IAX train has long since passed. I mean, if UncleKiwi finds it nice to use, more power to him, but those who do are few and far between |
| 13:28.31 | orn | Personally I'd try to solve it with SIP, and if I had NAT/FW issues I'd probably solve it with Wireguard |
| 13:29.03 | orn | But that's mostly just because I am so familiar with SIP and haven't used IAX more than just to test it over a decade ago |
| 13:29.56 | UncleKiwi | im having a read about wireguard |
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| 13:32.35 | seanbright | IAX is abandonware |
| 13:32.48 | seanbright | you should avoid it in new deployments |
| 13:33.50 | Samot | Agreed. |
| 13:34.36 | UncleKiwi | Samot: looks like I need to not use it :( thanks Samot for helping me become aware of that |
| 13:35.32 | UncleKiwi | it just seemed to easy to get up and running |
| 13:35.49 | UncleKiwi | i have it working on 1 other setup and its good |
| 13:35.54 | Samot | Getting SIP up and running isn't that hard either. |
| 13:36.26 | Samot | IAX is just something that doesn't solve any real problems these days. It lacks documentation, support and development. |
| 13:37.33 | seanbright | the big selling point for IAX2 when we used it was trunking, which saved us a shit-ton of bandwidth |
| 13:38.12 | UncleKiwi | thanks for the info and education. ok well im off to get some sleep almost 2am here |
| 13:38.31 | Samot | Yup, and back in the day when having 20Mbps was ultra super fast and expensive, it helped. |
| 13:38.37 | seanbright | indeed |
| 13:38.39 | Samot | Or even 10Mbps. |
| 13:39.54 | Samot | And yeah, even after all this time there are still plenty of router/firewall vendors who are very lax with their SIP handling. |
| 13:42.25 | Samot | orn: Fortigate would be one of those vendors. |
| 13:42.38 | Samot | WatchGuard is awful too. |
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| 14:27.19 | orn | i get cold shivers whenever a customer tells me they have a fortigate |
| 14:27.41 | seanbright | we use fortigate. wasn't too painful. |
| 14:28.04 | seanbright | "dear support: do don't want your device to touch our SIP traffic" |
| 14:28.10 | seanbright | "like... at all" |
| 14:29.00 | seanbright | i'm sure they exist, but i would just like a firewall firewall |
| 14:29.08 | seanbright | no IPS or deep packet inspection |
| 14:29.22 | file | moar fire |
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| 14:33.53 | orn | seanbright: Yeah -- that's pretty much what we do every time. But since we don't use Fortigate ourselves, we always have to figure out again how this is done and direct our customers what to do. |
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| 14:54.18 | InterLinked | So, for the dsp I want to pull in for my audiohooks-based notch filter, I don't need all the usual DSP stuff, like DTMF/MF/fax detection, progress detection, and so forth. |
| 14:54.20 | InterLinked | Would there be an issue with doing a subset of the stuff in __ast_dsp_new that I need to directly in my function file, or would I need to modify dsp.c for this? |
| 14:54.24 | InterLinked | https://github.com/asterisk/asterisk/blob/b4347c486150653ec7ce1d129e8f9017c69344da/main/dsp.c#L1696 |
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| 15:06.53 | orn | does anyone know if it's possible to get a channel's state within a confbridge to check whether or not it is muted with ARI? |
| 15:09.14 | orn | confbridge cli app is able to show this, but i haven't found a way to see this yet from within ARI |
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| 16:10.21 | igcewieling | We normally install a router and voice vlan so our phones don't go over the customer firewall. |
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| 17:56.46 | InterLinked | Is there currently no way to use the Asterisk DSP with a custom tone? I found ast_tone_detect_init() but that's currently only used for fax tones and isn't in dsp.h. Would dsp.c need a new function for this or have I missed something? |
| 17:57.44 | file | what you see is what there is |
| 17:57.50 | file | and you are in legacy old code that few people know or touch |
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| 18:04.08 | InterLinked | Okay, I was also looking for Goertzel algorithm related stuff, it seems like the stuff in dsp can identify a frequency but not filter it out, I guess Goertzel algorithm would need to be added? |
| 18:07.08 | file | I have no idea. |
| 18:09.54 | InterLinked | It does seem that ast_dsp_new initializes all the detectors, like fax, digit detection, and so forth. I guess I'd be best off adding a new entry function just for the custom tone. |
| 18:14.16 | avb | gentlemen, just wanted to confirm. If trunk requires you to use session timers, can that lead to the call without audio? |
| 18:14.57 | avb | i think that would lead to the call drop after the timer will expire, but doesnt seems to affect audio flow, right? |
| 18:15.31 | avb | trunk is using cisco as I see |
| 18:15.48 | avb | Sonus_UAC |
| 18:15.52 | file | session timers are independent of media flow |
| 18:16.21 | avb | thats what I thought |
| 18:16.38 | avb | ty @file |
| 18:17.15 | avb | still trying to win this dead-air situation |
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| 19:02.07 | igcewieling | The symptom of needing session timers and not having them enabled is the call goes silent or drops when the first session timer triggers, in my experience that is 10 mins into the call. |
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| 19:22.48 | sysgrammer_moe | hello all, I have setup DISA on asterisk but I still see my cellphone number as outgoing CID |
| 19:23.12 | sysgrammer_moe | I was under the impressing that outgoing CID would be of the ISP trunk |
| 19:23.16 | sysgrammer_moe | Please suggest |
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| 19:44.16 | igcewieling | sysgrammer_moe: you are using PRI connection to the telco? |
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| 21:10.58 | sysgrammer_moe | igcewieling: I am using a VOIP provider and not a analog card on my asterisk/freepbx machine if that answers your question? |
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| 22:19.34 | hitech95 | Hi guys I'm trying to use a pjsip trunk with IPV6. I'm getting a "Unsuitable transport selected (PJSIP_ETPNOTSUITABLE)" error. I have no idea why. The nameserver SRV record says it is a UDP connection: https://pastebin.freepbx.org/view/091691d6 |
| 22:33.17 | InterLinked | Is there something special that needs to be done for C++ source code? I ended up including a C++ file, but that didn't compile so well. I renamed my function to .cc, but get "No rule to make target". chan_vpb is C++ too but I don't see anything special in the makefile for it. |
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