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| 01:14.09 | tripleslash | I have a sip trunk provided via adtran from the provider so its sip but there's no nat. I'm having an issue where the external calls sound as if the person is talking through a fan. Its a very rapid stutter of sorts. The external caller doesn't have the problem, only the internal headset hears it. Internal to internal calls don't have the issue. |
| 01:19.49 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:19.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); 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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| 01:56.29 | igcewieling | sounds like a QoS issue. |
| 01:56.56 | igcewieling | tripleslash: use ulaw for everything when testing, it will save you a lot of sorrow. |
| 01:57.39 | tripleslash | igcewieling, it started out of the blue. I'm currently here by myself (ala almost 0 network traffic) and its still doing it. |
| 01:57.51 | tripleslash | I've already turned off all codecs but ulaw. |
| 01:57.59 | igcewieling | tripleslash: have you tried rebooting the box? |
| 01:58.09 | igcewieling | How is the Adtran connecting to the rest of the world? |
| 01:58.45 | tripleslash | yes. I've unpowered the adtran, the pbx, the switches (and thereby the phones) and its still the same. |
| 01:59.26 | igcewieling | How is the Adtran connecting to the rest of the world? T-1? EoC? |
| 02:00.06 | tripleslash | Adtran is via coax from provider. I setup a test twilio trunk so that I could bypass the adtran and do a call via our ISP but same end result. |
| 02:00.18 | tripleslash | internal to internal calls are fine. |
| 02:00.28 | igcewieling | what version of Asterisk? |
| 02:00.31 | tripleslash | Its only audio incoming to the org. Outgoing is fine. |
| 02:00.57 | tripleslash | ast 13.14.0 |
| 02:01.48 | igcewieling | I've needed coax connections on Adtrans. Do you have admin access to the adtran. |
| 02:01.49 | igcewieling | ? |
| 02:01.55 | tripleslash | no |
| 02:02.14 | tripleslash | its coax to cable modem to adtran. |
| 02:02.36 | tripleslash | Anything outside of the adtran is blackbox to me. They just provide all sip connections via single ip to my pbx's single ip. |
| 02:02.59 | tripleslash | Up until I setup the twilio trunk, I had nat completely turned off. |
| 02:04.45 | tripleslash | The pbx is in a vm but it has dedicated cpu cycles. I tried moving the vm to a different box just in case. I've tried adding cpus, taking the cpus down to 1 (in case of irq issues) and no change. |
| 02:08.19 | tripleslash | Started around March 2. |
| 02:29.23 | igcewieling | complain to your ISP. |
| 02:36.08 | tripleslash | except its happened via the sip provider and via data, so realistically, it shouldn't be isp, no? |
| 02:36.33 | Samot | The PBX is in a data center? |
| 02:36.51 | tripleslash | no |
| 02:37.07 | Samot | So its in the office with the phones? |
| 02:37.16 | tripleslash | yes |
| 02:37.31 | tripleslash | but the external connections are via two different data networks |
| 02:37.44 | Samot | Which are? |
| 02:46.23 | tripleslash | I haven't a clue how the sip is routed, only my data |
| 02:46.58 | tripleslash | they'll be out tomorrow to tell me everything looks good on their end and that its gotta be soemthing on my end and then I'll pull my last 3 hairs out and call it a day. |
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| 02:50.36 | Samot | What type of data connections are they? |
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| 03:08.53 | drmessano | lol |
| 03:09.50 | Samot | He took it to #freepbx because he asked there too. |
| 03:10.09 | drmessano | The problem is the same |
| 03:10.27 | Samot | I know. |
| 03:10.59 | drmessano | You know? |
| 03:11.18 | Samot | I might. |
| 03:11.24 | Samot | I'm not telling you first though. |
| 03:11.28 | Samot | So you tell me what you know. |
| 03:11.36 | Samot | Then I'll tell you if I know it too. |
| 03:12.02 | drmessano | In both cases it's still the ISP |
| 03:12.27 | Samot | The same ISP? |
| 03:12.47 | drmessano | The cable modem that's connected to the Adtran |
| 03:13.01 | drmessano | Is no different than the twilio test |
| 03:13.07 | Samot | Oh I know that part. |
| 03:13.13 | Samot | But there's two Internet pipes. |
| 03:13.18 | drmessano | So the issue is the cable provider |
| 03:13.30 | Samot | Twilio trunk has same issue over both connections... |
| 03:13.43 | Samot | Now if both connections are the same ISP and just two circuits..don't know. |
| 03:13.57 | drmessano | They're not the same ISP? |
| 03:14.08 | Samot | But one was described as cable and the other is the fiber network between the campus buildings. |
| 03:14.21 | Samot | How that fiber network gets out to the Internet, he doesn't know. |
| 03:14.27 | tripleslash | no no |
| 03:14.59 | tripleslash | two connections via cable, one is my data and goes into firewall and distributed. other is to adtran and then to asterisk. |
| 03:15.16 | drmessano | Right |
| 03:15.16 | tripleslash | cable modems are routed via two different networks outside by building afaik. |
| 03:15.38 | drmessano | The cable modems are on different ISPs? |
| 03:15.41 | tripleslash | data modem hasn't lost packets via long running pingplotter. |
| 03:16.19 | Samot | Are the data ISP and the voice ISP the same ISP? |
| 03:16.26 | drmessano | The cable modems are on different ISPs? |
| 03:16.28 | tripleslash | yes |
| 03:16.30 | tripleslash | no |
| 03:16.34 | drmessano | Ok |
| 03:16.38 | drmessano | Thats what I THOUGHT |
| 03:16.40 | Samot | So it's still Comcast. |
| 03:16.41 | Samot | OK |
| 03:16.44 | Samot | That does change thing. |
| 03:16.46 | tripleslash | buuuuuuut, when we have had data issues, voice was fine. |
| 03:16.47 | Samot | That does change things. |
| 03:16.53 | drmessano | I have the same fucking setup.. jeez |
| 03:17.13 | Samot | I have a client with the same. |
| 03:17.17 | Samot | It's the same connection. |
| 03:17.46 | Samot | Comcast drops one circuit.. |
| 03:17.51 | Samot | Splits it with two modems.. |
| 03:18.01 | Samot | Tells you that you now have "two" Internet connections |
| 03:18.03 | tripleslash | except two different sets of techs come out and two different support groups. The pri/sip people have said previously their sip isn't routed over the same network as the data. |
| 03:18.09 | drmessano | I dont know why there is any questions about "Well one is one the Adtran".. it's the cable. Ping tests are one, only slightly useful metric in this case |
| 03:18.09 | Samot | Because they each have their own IPs, etc. |
| 03:18.18 | drmessano | It's on the same network |
| 03:18.29 | Samot | ^^^^ |
| 03:18.31 | drmessano | It's the same Layer 2 |
| 03:18.36 | drmessano | It may not be the same Layer 3 |
| 03:18.39 | drmessano | But it's the same Layer 2 |
| 03:18.43 | Samot | Right. |
| 03:18.45 | tripleslash | I agree. |
| 03:18.58 | Samot | So that means if your actual INTERNET connection is having an issue... |
| 03:19.05 | Samot | both the modems are piping to it. |
| 03:19.06 | tripleslash | they're on different frequencies on the same coax |
| 03:19.17 | drmessano | ROFL |
| 03:19.21 | drmessano | Im sure they are |
| 03:19.30 | drmessano | That's so irrelevant |
| 03:19.33 | drmessano | Same node |
| 03:19.35 | drmessano | Same Layer 2 |
| 03:19.40 | drmessano | Same problems |
| 03:19.55 | Samot | Yup. |
| 03:20.06 | drmessano | There is nothing different about it |
| 03:20.30 | drmessano | If I put 2 DOCSIS 3 modems on the same node, they won't be on the same channels |
| 03:20.52 | drmessano | Whether it's residential, business, or business voice |
| 03:21.00 | drmessano | It's the same pipe |
| 03:22.58 | drmessano | The cable modem on the Adtran is only different in that it's on a different L3 network. That's all |
| 03:23.10 | drmessano | That's basically VLANing |
| 03:29.57 | drmessano | tripleslash: ^^^^ |
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| 03:44.22 | overyander | what is https://wiki.asterisk.org/wiki/display/AST/Advanced+pbx_lua+Topics referring to as "the main script" ? if i put db queries in various contexts, are all of the queries going to be ran for each channel regardless of the channel progressing through a context with a query? |
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| 06:18.06 | snadge | will asterisk accept a 44.1khz 16bit mono wav file? |
| 06:18.28 | snadge | my assumption was yes.. i ran it through our converter tool, and its dropped it to 8khz.. and that seems to work |
| 06:19.17 | snadge | but the original 44.1khz wav doesn't.. even though, as far as i can tell from the properties, they're identical besides the sampling rate |
| 06:23.59 | drmessano | Needs to be 8khz |
| 06:24.43 | drmessano | 8khz 16bit mono |
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| 07:00.26 | drmessano | So, how secure is SRTP? |
| 07:00.29 | drmessano | (j/k) |
| 07:01.53 | jkroon | haven't looked at the tech spec but based on what I've seen it should be fine as long as your phone does validation of the x509 cert for sip/tls. |
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| 07:12.27 | drmessano | Well nothing is really secure |
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| 07:42.27 | Samot | lol. |
| 07:44.24 | jkroon | burn it :) |
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| 14:25.51 | DanQuinney | Afternoon, I'm trying to use a Gosub in a custom dynamic feature but it doesn't appear to be working; http://paste.codebasehq.com/pastes/fc0nhzv7z1pgexgxcz |
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| 14:26.57 | DanQuinney | everything looks to be ok, it's almost like it doesn't support Gosub, but that's silly |
| 14:31.03 | DanQuinney | the docs suggest that Gosub does work |
| 14:34.57 | [TK]D-Fender | pauseMonitor => *100,self/caller,Gosub,"peers, pauseMonitor, 1" |
| 14:35.03 | [TK]D-Fender | Nver shove extra quotes |
| 14:35.13 | [TK]D-Fender | never shove SPACES in there either |
| 14:36.33 | DanQuinney | The docs say to use either of the following formats |
| 14:36.42 | DanQuinney | <FeatureName> = <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,"<AppArguments>"[,MOH_Class]] |
| 14:36.42 | DanQuinney | <FeatureName> = <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>([<AppArguments>])[,MOH_Class] |
| 14:38.15 | DanQuinney | ffs, it was the spaces?! |
| 14:38.28 | [TK]D-Fender | Spaces will finish you. |
| 14:38.38 | [TK]D-Fender | Don't get creative with spacing |
| 14:38.42 | DanQuinney | I'm off to drink vodka |
| 14:38.45 | [TK]D-Fender | This isn't liberal arts |
| 14:38.48 | DanQuinney | ha |
| 14:39.01 | [TK]D-Fender | extensions.conf was based on COBOL |
| 14:39.25 | [TK]D-Fender | Got it down to 26 bugs this morning. just fixed one so only 37 to go... |
| 14:40.03 | DanQuinney | jesus |
| 14:40.07 | DanQuinney | that's a lot |
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| 14:52.37 | ruied | I have compiled asterisk and now I don't have the "console" cmd. Is it something not set in menuselect before compilation? |
| 14:53.53 | ruied | I have a cronjob with: /usr/sbin/asterisk -rx "console dial D_D@script" |
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| 15:03.29 | [TK]D-Fender | probably due to missing an audio module for the call to use |
| 15:03.37 | [TK]D-Fender | chan_alsa, etc |
| 15:09.53 | ruied | yes, It seems to be that, installed alsa-dev and compiling asterisk now... |
| 15:11.15 | ruied | Is there any log or something that I could trace the cause of not having the "console" cmd ? |
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| 15:16.04 | ruied | it could return something back like: "Missing chan_alsa or chan_oss" or "error: no output channel..." |
| 15:16.21 | ruied | aouput/input |
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| 15:41.07 | ruied | The problem seems to be related to sound card channel. I had previously asterisk working in a ProxMox VM without sound card. I need the "console dial" but can't load chan_alsa because I do not have sound card. |
| 15:41.59 | ruied | It worked before and I'm not reminding of having issues with console. |
| 15:42.38 | ruied | is there a way to activate the console without the sound card? |
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| 15:45.40 | file | what are you needing to use it for without a soundcard? |
| 15:51.15 | adnauseum | from an ata syslog see the following 'T38 session started at port:1 ch:0, max rate:9600' question is what is controlling the max rate? Checked the ATA, no setting there for max rate |
| 15:52.38 | adnauseum | checked udptl_custom.conf and no max rate there either |
| 15:54.01 | *** join/#asterisk aiksa[LV] (5eba7827@gateway/web/freenode/ip.94.186.120.39) |
| 15:54.31 | aiksa[LV] | hey guys. |
| 15:54.41 | igcewieling | adnauseum: try #FreePBX |
| 15:54.42 | aiksa[LV] | having bit of a trouble with ARI here. |
| 15:55.31 | aiksa[LV] | I am trying to replicate Dial function; and instead of originate I use: channel create and then dial action to be able to passearly audio back to the initiating channel |
| 15:55.40 | igcewieling | res_fax.conf is where non-FreePBX Asterisk configures the max rate. |
| 15:56.02 | aiksa[LV] | however I cant seem to find way to change callerid for the outbound leg. |
| 15:56.21 | aiksa[LV] | the create channel and dial commands doesnt have callerid param |
| 15:56.39 | igcewieling | does it have a setvar param? |
| 15:56.46 | aiksa[LV] | so as i understand i would have to use channel setvar |
| 15:57.34 | aiksa[LV] | however no matter wheter i set other calllerid(num) on inititating channel (should have been copied to child, if i understand corectly) or on child right after channel create but befor bridge |
| 15:57.51 | aiksa[LV] | chan_sip still send invite with original callerid. |
| 15:58.26 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 15:58.48 | igcewieling | SIP doesn't really have CallerID, just stuff which looks like CallerID. First callerid(num) won't work. Try CALLERID(num) or even better _CALLERD(num) |
| 15:59.03 | igcewieling | _CALLERID(num) that is. |
| 15:59.38 | aiksa[LV] | with underscore at the beginnig? |
| 15:59.44 | rmudgett | igcewieling: Functions do not support channel variable inheritance. |
| 15:59.59 | igcewieling | rmudgett: Thanks. |
| 16:00.03 | rmudgett | For outgoing channels you need to set the CONNECTEDLINE() values. |
| 16:00.13 | igcewieling | aiksa[LV]: rmudgett pointed out my suggestion won't work. |
| 16:00.32 | aiksa[LV] | rmudgtett through setvar ? |
| 16:00.50 | *** join/#asterisk ruied (~ruied@81.84.234.209) |
| 16:00.55 | igcewieling | I thought CONNECEDLINE set the CallingID not CallerID? |
| 16:00.55 | aiksa[LV] | right after create =, but before dial right? |
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| 16:01.32 | aiksa[LV] | so, I should set CONNECTEDLINE() on initiating channel? |
| 16:01.58 | rmudgett | aiksa[LV]: See https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
| 16:02.06 | aiksa[LV] | Thanks! |
| 16:04.07 | aiksa[LV] | ok, the CONNECTEDLINE() is something else. This is to signal back the intiating party that the line has changed. |
| 16:05.01 | aiksa[LV] | setvar on CALLERID(num) does not work, neither does _CALLERID(num) |
| 16:05.45 | aiksa[LV] | It seems, that channel create command inherits values from the context of the channel it was called from |
| 16:05.50 | aiksa[LV] | is this assumption correct? |
| 16:07.25 | aiksa[LV] | wait a sec. i might be wrong about connectedline |
| 16:07.28 | aiksa[LV] | let me test |
| 16:07.58 | aiksa[LV] | It works! |
| 16:08.05 | aiksa[LV] | Many thanks! |
| 16:08.34 | aiksa[LV] | never thought of that in this context |
| 16:10.00 | *** join/#asterisk vince1 (~vince@ip72-202-187-29.ok.ok.cox.net) |
| 16:11.31 | aiksa[LV] | rmudgett: many many thanks! |
| 16:13.19 | aiksa[LV] | so the flow in short: initating channel recieved in stasis APP; new channel created, CONNETEDLINE(num) set on the newly created channel. Both of channels added to bridge. dial initiated on the newly created channel |
| 16:16.40 | Prelude2004c | hey guys.. anyone know if it is possible to transcode opus or codec2 > ulaw? |
| 16:19.53 | igcewieling | http://blogs.digium.com/2016/09/30/opus-in-asterisk/ |
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| 16:26.44 | drmessano | Is someone actually using Codec2 on phones? Why?! |
| 16:28.05 | igcewieling | maybe they think the audio quality of GSM isn't good enough? 8-| |
| 16:59.40 | igcewieling | heh, Verizon Wireless has announced they will no longer be issuing new IPv4 static IPs. |
| 17:02.12 | ruied | used chan_oss in my asterisk VM to make the "console dial"... |
| 17:04.37 | igcewieling | why do you need the console dial command? |
| 17:05.18 | igcewieling | It has been years since I've heard of anyone trying to use the server speakers and microphone as a gigantic softphone like you are trying to do. |
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| 17:32.52 | [TK]D-Fender | Using "console dial" ... on a system with no CONSOLE because of being a VM sounds incredibly backwards |
| 17:34.13 | igcewieling | I suspect he misunderstands the function of "console dial" but until he actually tells us WHY he wants to use it, it will forever remain a mystery. |
| 17:35.26 | [TK]D-Fender | Use a softphone and stop trying to treat * like a phone itself |
| 17:42.13 | igcewieling | I suspect he wants channel originate |
| 17:44.43 | [TK]D-Fender | I'm not even going to waste time on a guess |
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| 23:49.34 | topside | Iâm a SIP trunk provider working with a customer using Asterisk and theyâre having trouble getting a working config for our trunk which uses authentication but not registration. We challenge any unauthenticated INVITES and expect to receive authentication headers in the follow-up INVITE. Is there anything needed beyond just configuring the username/remotesecret that they need to do so that they respond to my 401 unauthorized? |
| 23:50.45 | WIMPy | no |
| 23:53.00 | snadge | so i asked yesterday about wav file format.. specifically the sampling rate, and was told it needs to be 8000hz |
| 23:53.12 | snadge | .. but don't some codecs support 16khz? |
| 23:53.20 | topside | Ok - didnt think so. He was raising a fit saying heâs never seen a SIP provider that requires authentication without registration and that he didnt think that could be configured with Asterisk :P |
| 23:53.30 | topside | or didnt know how to |
| 23:53.33 | WIMPy | 'core show file formats' |
| 23:54.11 | snadge | yeah thats cool.. i know that wav is supported, but specifically bit rates etc |
| 23:54.18 | WIMPy | Registration is not related to outgoing calls in any way. |
| 23:54.35 | topside | agreed |
| 23:54.36 | snadge | perhaps a 16khz file is supported.. i guess i could experiment with it, or even look at the code |
| 23:55.02 | WIMPy | .wav ist 8ks/s. if you want 16ks/s use .sln16. |
| 23:55.51 | topside | Iâm not an Asterisk guy, so just wanted to make sure that thereâs no issue with providing trunks to Asterisk users where authentication is used without registration. Thanks! |