IRC log for #asterisk on 20210514

00:10.03*** join/#asterisk andrewyager (~andrewyag@114.141.97.1)
00:35.37*** join/#asterisk darkdrgn2k (~darkdrgn5@unaffiliated/darkdrgn2k)
00:36.02darkdrgn2khi all, anyone know of any decent european sip providers. Looking for incoming calls specifically.
00:40.56sibiriainfobip is ok
01:56.02*** join/#asterisk mr44er1 (~mr44er@dynamic-046-114-007-090.46.114.pool.telefonica.de)
01:58.02*** join/#asterisk tsal (~tsal@i59F5F08B.versanet.de)
04:13.54*** join/#asterisk infobot (ibot@96-86-209-99-static.hfc.comcastbusiness.net)
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
06:01.53*** join/#asterisk Ner0Zer0 (~Ner0Zer0@87.253.63.54)
06:07.14*** join/#asterisk pchero (~pchero@211.178.226.108)
06:21.36*** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net)
06:37.48*** join/#asterisk lankanmon (~LKNnet@cpeb4fbe4e331bd-cm9050cadd5190.cpe.net.cable.rogers.com)
06:41.41*** part/#asterisk jbg (sid494673@gateway/web/irccloud.com/x-fnwhdhusgnlwtbaf)
07:01.09*** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir)
07:41.46*** join/#asterisk sinaowolabi (~Sina@102.134.114.1)
07:55.36*** join/#asterisk Ravenheart (Ravenheart@95-43-79-254.ip.btc-net.bg)
08:18.36*** join/#asterisk sinaowolabi (~Sina@102.134.114.1)
08:24.09*** join/#asterisk DodgeThis (~DodgeThis@246.102.90.149.rev.vodafone.pt)
08:44.33*** join/#asterisk kerouac[m] (kerouacmat@gateway/shell/matrix.org/x-advrgpyqmqyucijo)
09:10.53*** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi)
09:17.20*** join/#asterisk andrewya_ (~andrewyag@114.141.97.1)
09:22.48UncleKiwiHola
09:23.33UncleKiwifor some reason my dialplan is not showing when I type dialplan show - im using asterisk v18
09:47.29UncleKiwiok solved - a file from my dial plan was missing
10:02.22UncleKiwioh no - i tried to run voip over a wireless ISPs network connection and its giving audio that is choppy
10:07.55*** join/#asterisk sinaowolabi (~Sina@102.134.114.1)
10:13.44UncleKiwii 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
10:29.12UncleKiwiis choppy audio just bad news that is probably the internet connection and there is little i can do about it ?
10:29.29UncleKiwii have seen this situation before with this ISP
10:30.10UncleKiwimaybe the packets arrive out of order or slow or something - its just a mess
10:31.08UncleKiwipings report about a 50ms to the voip provider
10:31.21UncleKiwinormally 50ms is not an issue
10:33.13*** join/#asterisk sinaowolabi (~Sina@102.134.114.1)
10:38.28ornUncleKiwi: 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
10:41.24UncleKiwiorn: looks quite tricky to configure -
10:41.45UncleKiwido you mean like i should replace the internet connection
10:41.50UncleKiwior talk to the ISP
10:42.08UncleKiwisomething like that
10:43.22ornUncleKiwi: Are there no wired options available at all at this location?
10:44.05ornUncleKiwi: 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.
10:44.37UncleKiwiorn: 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
10:44.40ornbut yeah, something along those lines -- deal with the underlying problem by any means possible before trying to solve it with a jitter buffer
10:46.15ornFWIW, 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
10:47.13UncleKiwiits not over 4g its over some 5Ghz wireless maybe ubiquity
10:47.17UncleKiwihardware
10:48.23ornUncleKiwi: In my opinion, that is way worse, since that is an open spectrum liable to all sorts of interference
10:48.40ornis it a point-to-point connection with a directional antenna?
10:49.28UncleKiwiit would be point to multipoint - it will be a sector on a hill with abackhaul
10:49.45UncleKiwianyway yeah... its a fail for voip unfortunatly
10:50.10ornUncleKiwi: I see. It's very difficult to guarantee any sort of response rates on such connections. You might get all sorts of spikes.
10:50.22ornthe best you can hope for is an adaptive jitter buffer with a high maximum setting
10:50.44ornyou could just run ping for a while to see how high it spikes, and maybe use that as your max value
10:50.56ornbut you might be looking at quite the delay
10:52.06UncleKiwi:) it working
10:52.13UncleKiwiSet(JITTERBUFFER(adaptive)=700,,60)
10:54.37ornall righty -- if that works for you, great :) user's might experience some half-duplex radio type conversations though :)
10:54.52orn"hello, are you there, OVER" :)
10:56.02UncleKiwiim testing it again and again by calling the number and having ti play the tt-monty-knights
10:59.30UncleKiwiumm... yeah its not fixed
11:00.04UncleKiwiits just wireless and its moments of good conditions ie lack of congestion or interference
11:09.36filejitterbuffer only covers media from a channel, it doesn't help if it's media sent to a channel
11:09.43filefyi
11:10.06filein that case it is the responsibility of the receivers jitterbuffer to do things
11:12.43UncleKiwii 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 ?
11:14.19UncleKiwimaybe its just better i dont use this wireless connection for voip
11:24.58ornUncleKiwi: that would be optimal. i.e. not using the wireless connection. but as mentioned by file both sides need to use a jitter buffer
11:25.53fileThere are also situations where even a jitterbuffer can't help, as the buffering required just makes the experience poor
11:26.41*** join/#asterisk jess (jess@freenode/staff/jess)
11:27.05UncleKiwithanks for this info.
11:27.32UncleKiwiit seems to me that a quiality internet connection is key
11:28.37UncleKiwithe tt-monty-knights is quite an abusive audio - I have had a lot of farts in my general direction
11:33.47ornhahaha
11:33.50ornindeed
12:11.08*** join/#asterisk ghoti_ (~paul@dsl-rb-64-118-18-246.wtccommunications.ca)
12:16.14SamotIve run voip over WISPs
12:16.58*** join/#asterisk DodgeThis (~DodgeThis@246.102.90.149.rev.vodafone.pt)
12:33.14UncleKiwiHi Samot, did you have any trouble ?
12:33.41SamotIn general, no. Have I, yes. The quality of the WISP is always a factor.
12:33.44UncleKiwiit seems to work its just that the audio is choppy
12:33.58SamotWhat speeds does this provide?
12:34.13UncleKiwi30Mbit down 5Mbit up
12:34.28UncleKiwi50ms latency
12:34.45UncleKiwito the voip server
12:35.17SamotAnd every call is choppy?
12:35.25UncleKiwinah not all calls
12:35.40SamotSo how many calls?
12:36.01UncleKiwii am assuming its network congestion on the wireless network (ISPs network)
12:36.28UncleKiwii have just beeing calling the number and getting a test audio played
12:36.34UncleKiwii have probably called it maybe 50 times
12:37.04UncleKiwii just called it now and it was perfect
12:37.16SamotAnd out of those 50 times how many where choppy audio?
12:37.37UncleKiwimaybe 33%
12:37.58UncleKiwimaybe 50%
12:38.03UncleKiwiits better now
12:38.20UncleKiwitime here is amost 1am
12:38.33UncleKiwiso network is probably not in use
12:38.46SamotIs this a WISP network or mobile?
12:38.56UncleKiwiWISP
12:39.46UncleKiwii think thery are using 5g ubiquity air max
12:40.40UncleKiwii suspect its network congestion on the sector
12:41.03UncleKiwibecause as i test it now at 1am its perfect
12:41.25UncleKiwi12:41
12:42.18SamotOK so this really doesn't have to do with being wireless.
12:42.26SamotThis has to do with the ISP being oversold.
12:42.38SamotEasy to blame wireless.
12:42.49UncleKiwiprobably - aye
12:43.39UncleKiwii mean '
12:43.46SamotWell during "busy hours" you have choppy audio and probably other issues. Off hours, it's fine.
12:43.49UncleKiwi'normal' traffic at this time
12:44.04UncleKiwiit not normal traffic at this hour
12:44.17UncleKiwiyeah
12:44.39UncleKiwiim going to heat up a doughnut - just been to the supermarket
12:51.12UncleKiwii have a really fat customer and he twists up the telephone cord and destroys it
12:51.24UncleKiwiabout every 6 months
12:59.48UncleKiwiwhen using IAX2 to connect two asterisk PBXs by default its not encrypted - passwords would not be sent over the net in plain text right ?
13:01.15SamotShrug. I don't use IAX because, why.
13:01.40UncleKiwiIAX is useful
13:02.23UncleKiwifor connecting a couple offices together
13:02.42UncleKiwithats what i use it for
13:03.14UncleKiwiinternal calls and BLF can be passed thru the IAX
13:12.27SamotSame with SIP
13:16.00UncleKiwiyeah i think SIP can do it also
13:17.05UncleKiwibut IAX sends all the sigaling and audio all over one udp port - so rtp etc is not a concern
13:17.14SamotNo SIP can do it
13:17.28SamotOk. Those are 2005 issues.
13:18.00ornmeh... I still see a fair share of such issues -- people have old CPEs all over the place
13:18.17ornand even with newer ones (fortigate comes to mind)
13:18.26SamotOk
13:18.41ornbut there's other ways to remedy those, such as VPN or Wireguard etc
13:18.50ornUncleKiwi: you might want to look at a wireguard link between the pbxs
13:18.56SamotSo if this was such a huge problem why is IAX an Asterisk only option?
13:19.18ornwho said it was the only option?
13:19.31SamotI didnt
13:19.44SamotI asked why have more adopted it?
13:19.49SamotHavent
13:20.15orndid you?
13:20.21orni didn't see that
13:20.32SamotI asked why it was still an Asterisk only thing
13:20.57SamotIf SIP was so problematic and IAX solves said problems..why is SIP king?
13:21.45UncleKiwiIAX is used to join the two asteriusk boxes  - it can transport sip over the IAX link
13:21.55UncleKiwiwell sip calls
13:22.02UncleKiwithats what I do
13:22.15UncleKiwiIAX has its place and so does SIP
13:22.26UncleKiwiIAX is great for asterisk to asterisk
13:22.35SamotSo is SIP
13:22.46UncleKiwiok so use SIP
13:22.56SamotI do.
13:22.59UncleKiwiI find IAX more simple
13:23.18SamotBecause IAX doesnt solve any problems for me
13:23.21SamotNever has.
13:23.22UncleKiwiin the asterisk to asterisk connections
13:23.44*** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir)
13:25.08ornSamot: 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"
13:26.15ornSIP 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.
13:26.28SamotRight.
13:26.37SamotAnd IAX had an RFC draft
13:26.49SamotIt was in the protocol war
13:27.34ornYeah -- 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
13:28.31ornPersonally I'd try to solve it with SIP, and if I had NAT/FW issues I'd probably solve it with Wireguard
13:29.03ornBut 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
13:29.56UncleKiwiim having a read about wireguard
13:30.01*** join/#asterisk pchero (~pchero@211.178.226.108)
13:32.35seanbrightIAX is abandonware
13:32.48seanbrightyou should avoid it in new deployments
13:33.50SamotAgreed.
13:34.36UncleKiwiSamot: looks like I need to not use it :(  thanks Samot for helping me become aware of that
13:35.32UncleKiwiit just seemed to easy to get up and running
13:35.49UncleKiwii have it working on 1 other setup and its good
13:35.54SamotGetting SIP up and running isn't that hard either.
13:36.26SamotIAX is just something that doesn't solve any real problems these days. It lacks documentation, support and development.
13:37.33seanbrightthe big selling point for IAX2 when we used it was trunking, which saved us a shit-ton of bandwidth
13:38.12UncleKiwithanks for the info and education. ok well im off to get some sleep almost 2am here
13:38.31SamotYup, and back in the day when having 20Mbps was ultra super fast and expensive, it helped.
13:38.37seanbrightindeed
13:38.39SamotOr even 10Mbps.
13:39.54SamotAnd yeah, even after all this time there are still plenty of router/firewall vendors who are very lax with their SIP handling.
13:42.25Samotorn: Fortigate would be one of those vendors.
13:42.38SamotWatchGuard is awful too.
13:52.35*** join/#asterisk InterLinked (~ambassado@cpe-24-209-155-151.wi.res.rr.com)
14:07.56*** join/#asterisk sinaowolabi (~Sina@169.159.95.219)
14:27.19orni get cold shivers whenever a customer tells me they have a fortigate
14:27.41seanbrightwe use fortigate. wasn't too painful.
14:28.04seanbright"dear support: do don't want your device to touch our SIP traffic"
14:28.10seanbright"like... at all"
14:29.00seanbrighti'm sure they exist, but i would just like a firewall firewall
14:29.08seanbrightno IPS or deep packet inspection
14:29.22filemoar fire
14:31.33*** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-xxpvfhilcciraohk)
14:31.33*** mode/#asterisk [+o bford] by ChanServ
14:33.53ornseanbright: 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.
14:34.49*** join/#asterisk HannaM (~quassel@p54849510.dip0.t-ipconnect.de)
14:39.10*** join/#asterisk gschanuel (~gschanuel@200-181-252-244.user3p.brasiltelecom.net.br)
14:43.39*** join/#asterisk sinaowolabi (~Sina@169.159.67.69)
14:48.19*** join/#asterisk BakaKuna (~Thunderbi@2a02-a446-ae46-1-8e50-259-47a4-6d14.fixed6.kpn.net)
14:54.18InterLinkedSo, 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.
14:54.20InterLinkedWould 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?
14:54.24InterLinkedhttps://github.com/asterisk/asterisk/blob/b4347c486150653ec7ce1d129e8f9017c69344da/main/dsp.c#L1696
14:56.53*** join/#asterisk rpifan (~rpifan@p200300d2670b9500982c0fd39c3471d7.dip0.t-ipconnect.de)
14:59.31*** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772)
15:06.53orndoes 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?
15:09.14ornconfbridge cli app is able to show this, but i haven't found a way to see this yet from within ARI
15:11.10*** join/#asterisk overyander (~overyande@216.163.21.11)
16:00.17*** join/#asterisk CatCow97 (~mine9@c-73-96-109-206.hsd1.or.comcast.net)
16:04.05*** join/#asterisk waldo323__ (~waldo323@d149-67-45-83.clv.wideopenwest.com)
16:10.21igcewielingWe normally install a router and voice vlan so our phones don't go over the customer firewall.
16:45.13*** join/#asterisk sinaowolabi (~Sina@102.134.114.1)
17:02.30*** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1)
17:12.49*** join/#asterisk rpifan (~rpifan@p200300d2670b950094429430374addbc.dip0.t-ipconnect.de)
17:56.46InterLinkedIs 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?
17:57.44filewhat you see is what there is
17:57.50fileand you are in legacy old code that few people know or touch
18:02.18*** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net)
18:04.08InterLinkedOkay, 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?
18:07.08fileI have no idea.
18:09.54InterLinkedIt 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.
18:14.16avbgentlemen, just wanted to confirm. If trunk requires you to use session timers, can that lead to the call without audio?
18:14.57avbi think  that would lead to the call drop after the timer will expire, but doesnt seems to affect audio flow, right?
18:15.31avbtrunk is using cisco as I see
18:15.48avbSonus_UAC
18:15.52filesession timers are independent of media flow
18:16.21avbthats what I thought
18:16.38avbty @file
18:17.15avbstill trying to win this dead-air  situation
18:28.55*** join/#asterisk post-factum (~post-fact@vulcan.natalenko.name)
18:30.05*** join/#asterisk sinaowolabi (~Sina@169.159.67.69)
19:02.07igcewielingThe 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.
19:22.05*** join/#asterisk sysgrammer_moe (~sysgramme@d50-117-157-138.yt.northwestel.net)
19:22.48sysgrammer_moehello all, I have setup DISA on asterisk but I still see my cellphone number as outgoing CID
19:23.12sysgrammer_moeI was under the impressing that outgoing CID would be of the ISP trunk
19:23.16sysgrammer_moePlease suggest
19:41.01*** join/#asterisk rpifan (~rpifan@p200300d2670b95001f2727383ebdc686.dip0.t-ipconnect.de)
19:44.16igcewielingsysgrammer_moe: you are using PRI connection to the telco?
20:06.45*** join/#asterisk ghoti (~paul@dsl-rb-64-118-20-57.wtccommunications.ca)
21:10.58sysgrammer_moeigcewieling: I am using a VOIP provider and not a analog card on my asterisk/freepbx machine if that answers your question?
22:16.07*** join/#asterisk Typhon (~Typhon@dslb-088-066-010-246.088.066.pools.vodafone-ip.de)
22:16.44*** join/#asterisk hitech95 (33b7aa84@51.183.170.132)
22:19.34hitech95Hi 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
22:33.17InterLinkedIs 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.
22:52.11*** join/#asterisk akp55 (~akp55@pool-71-115-0-190.rcmdva.fios.verizon.net)
22:54.18*** join/#asterisk akp55 (~akp55@pool-71-115-0-190.rcmdva.fios.verizon.net)
22:56.08*** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir)
23:35.38*** join/#asterisk InterLinked (~ambassado@cpe-24-209-155-151.wi.res.rr.com)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.