IRC log for #asterisk on 20210106

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11:14.35allizomHi all, I'm trying to configure asterisk to receive calls from my provider, but I can't still grasp how do I let calls from any number in. The error I receive is "No matching peer for '0039PHONENUMBERCALLINGME' from '213.205.21.8:5060'" and here are config and logs: https://paste.centos.org/view/raw/ac3072c8
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13:56.52Kobazallizom: you haven't configured an endpoint that matches either the username or the ip address that it's coming from
13:58.02Kobazif the carrier is sending From: <PhoneNumber> in the sip headers, then you'll want to do an ip match
13:58.27Kobazallizom: dpaste your pjsip set logger on, output
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14:05.39SamotKobaz: How will the pjsip logger help in all this?
14:07.06KobazYou'll see exactly how the call is coming in, so you can configure the endpoint to match
14:07.29Kobazwe can assume certain things, but it's good to know for sure
14:07.51Kobazoh, that might be a chan_sip error
14:07.56Kobazi didn't look at the logs yet actually
14:08.03SamotRiiight
14:08.09Kobazsorry, haha, it's morning
14:08.12Kobazhe's got all the logs in there
14:08.18SamotSure.
14:08.21Kobazallizom: never mind, i didn't have coffee yet
14:10.03Kobazallizom: insecure=invite .. your ITSP is not sending you any authentication and you want to match based on the ip.   and you have a possible issue.. dns lookup ims.tiscali.net results in an empty response... are you using private dns?
14:10.06allizomKobaz, Samot: thanks, I dug into my issue and with this configuration I've been able to receive calls: https://paste.centos.org/view/raw/1ba52f84 - I still can't place them probably because my To: header is not set to the requested one as you can see in the log: https://paste.centos.org/view/raw/ff222b6e - Yes, it's chan_sip due to my provider using tel uris
14:10.47allizomKobaz: ims.tiscali.net is not resolvable via A record - but nevermind that
14:12.03SamotWhat does tel uris have to do with chan_sip?
14:12.24allizomSamot: AFAICT pjsip does not handle tel uris
14:13.06allizomat least in the snowflake version used by my provider
14:13.14SamotUhm..
14:14.43SamotYour provider doesnt care about chan_sip vs chan_pjsip
14:14.44Kobazallizom: I see the cancel... so your carrier is stopping the call after proceeding
14:15.03Kobazsession progress and then cancel
14:15.18allizomSamot: they don't, it's just that if I use pjsip I can't handle tel uris they use/require
14:15.51allizomKobaz: yes, they probably want my To: header to contain ims.tiscali.net
14:15.52SamotThey use it in the from header
14:16.19SamotNot in the request header where asterisk gets the DID from
14:16.36Kobazallizom: well... asterisk is sending the cancel
14:16.49KobazReliably Transmitting (no NAT) to 213.205.21.8:5060:   CANCEL sip:PHONENUMBERIMCALLING@213.205.21.8 SIP/2.0
14:17.53SamotCancel came from grandstreAm
14:18.10Kobazi didn't get that far yet, but yeah it's passing it through
14:18.11SamotThen from asterisk to the provider
14:18.24SamotSo the call was hung up
14:18.26fileoutgoing call to provider was placed, the dialed number was not found (and is likely being said in session progress), caller hears this and hangs up
14:18.39allizomfile: correct
14:18.46filebows
14:19.10Kobazdo you have an audio recording?
14:19.20Kobazis it saying 'call cannot be completed as dialed' or some such
14:19.42Kobazand if so, contact your carrier and ask, umm, why?
14:19.43allizomit said something like "the dialed number does not exist"
14:20.02KobazI've had this problem with verizon.  They are VERY picky.  You need to pass P-Asserted-Identity with your BTN in order to make calls
14:20.18Kobazthey will send you early media that the dialed number cannot go through, otherwise
14:21.05SamotOr perhaps they want calk
14:21.18SamotCallerid not in the from header
14:21.47Kobazallizom: do you have a username credential for your carrier?
14:22.06KobazMYPHONENUMBER@ims.tiscali.net  is that your username you're supposed to use?
14:22.14allizomKobaz: it is identical to my full phone number
14:22.21allizomyes
14:22.28Kobazis your username phonenumber... or phonenumber@...the whole thing
14:23.33Kobazyou'll want fromuser=MYPHONENUMBER@ims.tiscali.net  assuming you need that for your username
14:23.49Kobazand then probably, sendrpid=yes to pass callerid
14:23.57allizomlet me try this
14:25.37SamotOr PAI.
14:25.43SamotThat is a question for the ITSP.
14:25.58SamotWhat format do they want CallerID in?
14:26.12SamotDon't assume RPID, a failed RFC, is what is being used.
14:26.42Kobazmany, many systems support this, it's pretty common that it will work
14:27.13Kobazin my experience PAI actually works less often, but yeah, either one is a consideration
14:27.41SamotOr perhaps they want "P-Called-Party-ID" which is what they use to send incoming callerid
14:27.49SamotIn your experience?
14:27.56SamotPAI is an RFC standard
14:28.04SamotRPID is a failed RFC
14:28.27Kobazreally depends on the carrier
14:28.33SamotExactly
14:28.33Kobazor the device you're talking to
14:28.41*** join/#asterisk tips (~tips@pool-173-72-12-154.cmdnnj.fios.verizon.net)
14:28.45SamotHence my statement of "This is a question for the ITSP"
14:28.49Kobazsure
14:28.53allizomHow do I set To: header to be <sip:PHONENUMBERIMCALLING@ims.tiscali.net> rather than <sip:PHONENUMBERIMCALLING@213.205.21.8> ?
14:29.59SamotYou set the host as that
14:30.17Kobazhttps://dpaste.com/8CGZR9A7P
14:30.24SamotIt's using the host setting to set the 213.205.21.8
14:30.30Kobazthat too
14:31.05KobazSamot: but his problem, is that ims.tiscali.net doesn't resolve
14:31.11SamotNo.
14:31.22Samotims.tiscali.net is probably  a SRV record.
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14:32.00Kobaznot sure, it doesn't resolve any which way for me
14:33.49Samotallizom: What has the ITSP said?
14:33.55allizom"host -t SRV ims.tiscali.net" from my network gives "ims.tiscali.net has no SRV record"
14:34.15Kobazwhy is your carrier giving you a dns with no records?
14:34.41SamotMost likely for routing/account purposes.
14:34.56Samotallizom: What has the ITSP said?
14:35.09SamotYou've contacted them about not being able to make outbound calls?
14:35.25SamotThey've looked at an attempt and told you "X, Y, Z is incorrect"
14:35.46*** join/#asterisk AsteriskRoss (~AsteriskR@37.157.48.2)
14:35.48allizomthey provided this one: core1.p.ims.tiscali.net which resolves to that IP address but is not identical to ims.tiscali.net
14:35.58SamotSore?
14:35.59SamotSo?
14:36.04SamotUse what they provide you
14:36.04Kobazthey provided that to do what with?
14:36.14allizomSamot: I can make calls without using asterisk, their service sorta works
14:36.18KobazUse as the proxy? the From: header? the what?
14:36.41*** join/#asterisk AsteriskRoss (~AsteriskR@37.157.48.2)
14:36.43allizomlet me read again the info they provided me
14:36.45KobazIf their service soft of works with a soft phone, why would you configure asterisk without asking the carrier, "why does this kind of work?"
14:37.31SamotMy guess, that's the OB Proxy
14:38.17allizomusername: MYPHONENUMBER@ims.tiscali.net, outbound proxy ip address: 213.205.21.8
14:38.40allizomdomain/registrar: ims.tiscali.net
14:38.42SamotSo that would mean fromuser=MYPHONENUMBER
14:38.53Samotfromdomain=ims.tiscali.net
14:39.03Samotoutboundproxy=213.205.21.8
14:39.16SamotSo where does this core1.p.tiscali.net come into play?
14:39.22SamotWhat did they give that to you for?
14:39.51allizomit's just an alternate way they told me to find their outbound proxy
14:39.57SamotOK
14:39.58allizomthey specify both
14:40.00Kobazalternate way?
14:40.10Kobazsounds like the only way?  if the other domain has no dns records
14:40.27allizomcore1.p.tiscali.net resolves to 213.205.21.8
14:40.36allizomso just use the name or the ip address directly
14:40.50SamotSo you lack the outboundproxy setting
14:40.55KobazNames are good, in case they decide to change this own the line
14:41.16allizomSamot: I'm adding it
14:41.36allizomwith host=ims.tiscali.net
14:44.41allizomSamot: no luck. with host=ims.tiscali.net and outboundproxy=213.205.21.8 and fromuser=00390924507049 I'm back to not even being able to receive calls
14:44.55SamotHeh I figured.
14:45.01allizomsip show peers: Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
14:45.01allizomitsp                      (Unspecified)
14:45.09Kobazright
14:45.13SamotYou need to have two peers most likely
14:45.20SamotAre you in Germany?
14:45.29allizomnope, IT
14:45.35SamotClose enough.
14:45.40Kobazheh
14:45.45SamotThose EU countries love themselves IMS
14:46.04SamotAnd using the old school incoming and outgoing users.
14:46.19SamotBasically incoming calls are going to match based on the host= setting
14:46.35SamotSo whatever is in the host= setting needs to resolve to an IP if it is a FQDN
14:46.55SamotIt's also the setting used for the domain part of Request/To headers.
14:47.11SamotThis should be a Chan_PJSIP trunk to avoid all this BS.
14:48.35allizomSamot: I've tried and also read in the bug tracker, that I can't use that unfortunately due as said to those tel uris my provider uses
14:48.45Kobazthat doesn't make sense
14:48.50Kobazwhy would chan_sip work any better
14:49.02Kobazit has way less flexability with regards to that
14:49.19Kobazcan you link one of the bugs?
14:49.25allizomyes, one minute
14:49.29filechan_pjsip doesn't support Tel URIs as noone has put in the time to audit and add all the support for it
14:49.41fileURIs in PJSIP are parsed into separate structures and you have to explicitly treat and use them differently
14:49.47KobazOh, good to know. Never had to deal with tel:
14:50.07filetreat it as a SIP URI and kaboom, you crash
14:50.34KobazWell okay then! hah
14:50.38Kobazchan_sip it is
14:51.33allizomso my question stands, I need to use host=213.205.21.8 *and* to send To: headers with ims.tiscali.net, is this possible some way?
14:52.03Samot9:45:13 AM <Samot> You need to have two peers most likely
14:52.07Kobazhttps://dpaste.com/8CGZR9A7P
14:52.21KobazAnd to alter the To: or the From: header, you can additionally append the following to any of the above strings:
14:52.32Kobazset up your peers with proper ips
14:52.38Kobazand then modify it run-time
14:53.30allizomKobaz: I'm currently calling it this way: Dial(SIP/${EXTEN}@itsp)
14:53.41allizomso, let me see..
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14:54.28allizomit should be the second case: SIP/username@domain   (SIP uri)
14:55.25allizomcan I specify todomain without touser? I guess I should write touser anyway, with a variable
14:58.57allizomI'm going to try with Dial(SIP/${EXTEN}@itsp!${EXTEN}@ims.tiscali.net)
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15:05.47allizomSamot: while it did in fact send the To: header I intended, I still can't place calls. How should I configure those two peers?
15:06.05Kobazshow your new debugs?
15:07.50allizomof course, let me save them
15:13.57allizomhere it is: https://paste.centos.org/view/raw/08e686bb
15:16.36SamotOK
15:16.40SamotI'm going to ask this again
15:16.52SamotHave you asked the ITSP why they are rejecting your calls?
15:17.28KobazYeah, now's definitely the time to do that
15:17.56Kobazbecause the headers look fine now (according to what I believe you should be sending (based on your limited information from your carrier))
15:18.32allizomnope, I can try and hope for the best, but I really think I'll be unable to talk with someone knowledgeable enough. I'll do that though
15:19.27SamotGD.
15:19.34SamotFind a better provider then.
15:19.58allizomThank you. This is what I can get right now unfortunately
15:20.05KobazYeah, if your carrier can't answer a basic question on 'why did this fail?', then move on
15:20.12allizomBut I will talk to them
15:20.18SamotSorry but I'm just sick of that excuse.
15:20.20SamotOr "reason"
15:20.24allizomIf all else fails, I'll take a log from my ATA
15:20.29allizomand compare the two
15:20.35SamotEveryone seems to always have a provider that is worse at this then them.
15:20.49KobazSamot: haha true
15:21.05Kobazallizom: the ata is working?
15:21.24allizomyes, if I connect directly with my provider credentials
15:21.29Kobazsdkfjhasdfhjaskhfasdf
15:21.34Kobazand you waited until now to share this?
15:22.04allizomwell, yes?
15:22.05KobazYou realize there is 100% less guessing involved if you have a working configuration to go against
15:22.33allizomthat's why I said I want to compare the two
15:22.38KobazThe last half hour was a complete waste then
15:22.44allizomoh, sorry
15:22.48KobazIt's all just guessing at what's supposed to work
15:23.08allizomI see
15:23.10KobazKnowing for sure, is the way to go
15:23.23Kobaz"This works"  Lets replicate this to another system
15:23.27Kobazversus starting from scratch
15:24.03KobazSo yeah. show a full call via the ATA, and then that'll be the roadmap on what to have asterisk send
15:24.51allizomit's unfortunately going to take a while, but I'm going to do exactly that
15:24.59KobazYeah, that's the best way
15:25.06KobazThen you know what it's supposed to look like
15:25.17KobazThat's literally my motto at the office.  "We need to know 'for sure'"
15:25.30KobazNo guessing, or, as little as possible
15:26.25Samot🍿
15:27.05SamotThis is awesome.
15:27.27allizomoh, actually I told you: 15:36:14 - allizom: Samot: I can make calls without using asterisk, their service sorta works
15:27.36SamotKobaz: You literally joined this conversation with assumptions and guesses before even looking at the provided data.
15:27.40Kobazallizom: you didn't provide many details
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15:27.56seanbrightSamot: STOP ARGUING WITH KOBAZ
15:28.00Kobazhaha
15:28.03seanbrightthis is your SECOND WARNING
15:28.14KobazSamot: My brain skipped over the url, that's not my normal modus
15:28.40seanbrightis just kidding, hopefully that is obvious
15:28.42KobazSamot: as little guessing as possible... and... based on what information I had... well.. that was the only start
15:29.59Kobazallizom: and i asked in response: .... [ask] the carrier, "why does this kind of work?"
15:30.09SamotKinda work?
15:30.12SamotOK.
15:30.16SamotATAs, softphones...
15:30.21Kobaz'sorta works' 'kind of works'
15:30.24allizomwas not allocating blame :P
15:30.24SamotThey kinda have a set purpose..
15:30.27Kobazthat's when you ask the carrier
15:30.38SamotThey have nice fields in their GUI for you to put details in.
15:30.48SamotAsterisk is a telephony kit.
15:31.01KobazWhat doesn't work about it?
15:31.10SamotKinda got to do all those configs that the ATAs and softphones already have in place.
15:31.18SamotSo when it works on an ATA vs Asterisk...
15:31.27Kobazallizom:  What's the 'sorta' part?
15:31.34SamotIt either means Asterisk is configured wrong or they are blocking PBXes on the service.
15:31.52KobazYou can always set the user agent
15:31.53allizomthat it uses tel uris in a particular, nonstandard way
15:32.21KobazHi, I'm a grandstream
15:32.26KobazHi, I'm a mac
15:39.48*** join/#asterisk allizom1 (~Thunderbi@unaffiliated/allizom)
15:42.01allizom1This is the log of a successful call I placed: https://paste.centos.org/view/raw/a124191a
15:44.06Kobazthat looks like wireshark
15:44.16allizomyes, sorry it's rather unwieldy
15:44.21Kobazcan you upload the pcap instead
15:44.34KobazIt's a little easier to see when it's unformatted
15:44.57allizomI know, I just don't have a place to upload it to
15:44.57Kobazbecause then you see the actual headers as they are, literally
15:45.21allizommaybe there's a way to just extract the sip part
15:45.25SamotThat was made straight from the Grandstream?
15:45.28Kobazhttps://gofile.io/ or such
15:45.36SamotWhere was this capture done at?
15:45.56allizomit's the capture from the ATA taken from my home router
15:46.22SamotSo again
15:46.28SamotEither Asterisk is configured wrong
15:46.32allizomI'm trying to upload it somewhere
15:46.36SamotOr they are blocking the use of Asterisk
15:46.50SamotCall. The. ITSP.
15:46.53SamotJFC.
15:47.12allizomok, I will, as I said before
15:47.19SamotI know, i know.
15:47.23KobazThe main things I get out of this: P-Preferred-Identity    and  <number>@ims.tiscali.net
15:47.26SamotThey know less than you.
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15:47.57allizomshrugs
15:48.21SamotAgain, either Asterisk is configured wrong and they need to tell you what to correct.
15:48.36SamotOr they are blocking the use of it.
15:48.49Kobazwell you can figure out what needs to be corrected based on this dump vs one that's not working
15:48.56Kobazincluding the user agent
15:49.07Kobazif that's causing a block or whatever
15:49.49allizomKobaz: I guess if they were actively blocking that I would not be able to even receive calls, but. Still a guess
15:50.22allizomcall them and figure out someway. That's fine
15:50.28allizomthank you all
15:52.36KobazNow you have some tools to figure this out, and... how to get better help in the future
15:52.40KobazYou're welcome
15:52.49allizomKobaz: point taken
15:57.32Kobazhere's what you can do
15:57.45Kobazallizom: take the pcap from the ata
15:57.58Kobazuse tcpdump to filter out only the ITSP traffic.. use host xxxx, etc
15:58.22Kobaztcpdump -r <pcapfile> host 213.205.21.8 -A -s0 > /tmp/working
15:58.29Kobaztake a pcap on the asterisk box
15:58.37Kobaztcpdump -r <pcapfile> host 213.205.21.8 -A -s0 > /tmp/notworking
15:59.00Kobazand throw them in a diff tool.  Line everything up in terms of the INVITEs and see what needs to be changed
16:06.06SamotOr call support
16:06.59Kobaz*and call support
16:07.01Kobazand be like wtf
16:07.40allizomI'll do both. This is one: https://paste.centos.org/view/raw/2e54f539 and tomorrow (today it's holiday) I'll call them
16:08.07seanbrightrport
16:08.42seanbrightthat's the first difference that jumps out at me
16:09.06seanbrightactually the request URI is different too
16:09.19Kobazand Route: is missing on the 'not working'
16:09.22SamotThe easy fix
16:10.00SamotAdd the domain and IP to /etc/hosts
16:10.27SamotSo it will resolve locally to the right IP
16:10.30Kobazworking as P-Preferred-Identity, notworking, does not
16:10.45SamotThe ATA isnt trying to resolve it
16:11.08SamotIts resolving the ob procy
16:11.13SamotProxy
16:11.15Kobazallizom: sort the fields alphabetically in each INVITE. and go line by line, and get the appropriate setting in asterisk
16:11.51SamotAgain
16:12.10KobazAnd if that doesn't work, definitely something going on with the carrier
16:12.15SamotThe ATA is not trying to resolve the host domain
16:12.20SamotAsterisk is
16:12.26SamotNo
16:12.31SamotJfc
16:12.41seanbrightheh
16:13.22*** join/#asterisk AsteriskRoss (~AsteriskR@37.157.48.2)
16:14.09seanbrightKobaz: Samot: we're going to play roshambo to see who helps allizom: please PM your choice of one of the following: rock, paper, scissors
16:14.23SamotNaw
16:14.29SamotIts all Kobaz
16:14.39seanbrightdon't give up
16:15.07allizomeasy people, I can try on my own :)
16:15.08SamotMeh. Im in negative fucks
16:15.09KobazIt definitely can work.  Samot's all trying to throw in the towel so soon
16:15.22SamotYeah that is it
16:15.25seanbrightKobaz sent me 'Apricots' via PM so instructions must have been unclear
16:15.36Kobazhaha
16:16.08KobazI've missed you guys
16:16.23KobazI literally took like a 4 year hiatus from active participation in #asterisk
16:16.36KobazBeen so friggin busy
16:17.56allizomSamot: if I add "213.205.21.8 ims.tiscali.net" to /etc/hosts do I need outboundproxy= and fromdomain= anymore ?
16:18.10SamotYes you do
16:18.14allizomok
16:18.26SamotThat just solves dns resolution
16:18.49SamotBecause Asterisk is doing a lookup on the domain
16:18.57SamotThat doesnt have a record
16:19.13SamotThe grandstream is not
16:19.24SamotIts using the ob proxy
16:20.29allizomSamot: that was it! I've succesfully placed a call right now
16:20.45allizomoh goodness
16:20.45Kobazyay
16:20.51seanbrightnice. i knew Kobaz would get it resolved...
16:21.05KobazAnd... you guys... I did point this out 25 minutes ago, that the domain had no dns
16:21.12SamotIts like ATAs were configured a certain way
16:21.20SamotI know
16:21.51SamotIm explaining why the ATA worked and Asterisk did not
16:22.05SamotAsterisk wasnt configured right
16:22.07Kobazseanbright: the propos has to be to allizom for sticking around so long
16:22.52allizomI can be stubborn if I want to
16:22.57allizomwhich is not always a pro
16:23.28SamotI dont allow pure IP requests either
16:23.48Kobazright, avoids a lot of crap calls coming in
16:24.04SamotAnd that would have been the answer if you called my support
16:24.15SamotAnd you were my user
16:24.28Kobazbut you're off on holiday today
16:24.36SamotNo im not
16:24.42Kobazi'm just saying, if you were the carrier
16:24.49SamotIm on call 24/7/365
16:24.58Kobazyou completely missed it
16:25.01SamotIm a carrier
16:25.08SamotI have holiday support
16:25.16Kobaz[2021-01-06 11:07:40] <allizom> ..... and tomorrow (today it's holiday) I'll call them
16:25.24SamotI know
16:25.36KobazI'm giving you a hard time, you know
16:25.38seanbrightso friday is my birthday
16:25.42SamotSo the carrier lacks holiday support?
16:25.46KobazHappy astribirthday
16:26.00KobazYeah that's crazy, what carrier is actually closed... ever?
16:26.07seanbrighti hope that you've sent gifts via UPS/FedEx as USPS has been pretty slow
16:26.34KobazBTC can make it over the same day
16:26.45seanbrightsure, i'll take 1
16:27.04Kobazi just about trippled my money lately
16:27.38KobazNeed to set a good sell trigger when the impending crash occurs
16:30.48Kobazallizom: how's your tel: uris working
16:31.29allizomKobaz: they send it on inbound calls
16:31.37allizomand you still need to handle that
16:31.46KobazI need to handle that?
16:32.05allizomyes, 'you'
16:32.08Kobazhaha
16:32.12Kobaz:P
16:32.14allizomfor some values of 'you'
16:34.34grummundHi, what's the right way to replace the default voicemail greeting?
16:35.02Kobazyou can A) replace the wav/etc files in the sound path, or B) write your own voicemail
16:35.23seanbrightgrummund: "comedian mail, mailbox?" you mean?
16:35.57grummundreplacing the file /usr/share/asterisk/sounds/en_US_f_Allison/vm-intro.wav works but seems a bit "hacky", so just checking on the right way.
16:36.19Kobazseanbright: I just have mine say 'Mailbox?'
16:37.26seanbrighthttps://github.com/asterisk/asterisk/blob/master/CHANGES#L1280-L1283
16:37.30seanbrightgrummund: ^^
16:37.47grummundit is presumably comedian mail, whatever is the stock VoiceMail() app.
16:40.27grummund"Please leave your message after the tone, when done hangup or press the pound key."
16:40.39grummundsorry, i mean that message ^
16:42.16seanbrightjust scanning the sample voicemail.conf i don't see it
16:43.17grummundthe stock file is vm-intro.wav and just replacing works, but i guess might be clobbered on a upgrade.
16:44.37seanbrightyeah, looks like you may have to do that
16:45.16grummundthat's fine.
16:59.36grummundjust to be sure... there are no free British accent voice prompts for asterisk, right?
17:19.54seanbrighti could record some for you
17:20.05seanbrightg'day govna
17:20.21seanbrightpop pop cherrio
17:20.27seanbrightcheerio*
17:20.40grummundyeah i found some good ones like that ;)
17:21.06fileseanbright: you doing a British accent... that would be... something
17:21.41seanbrightput another shrimp on the barbieeeeee
17:22.38grummundhttps://ymstat.com/dyn/community/13012_orig.mp3
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19:14.21cloud9hey everyone. I have asterisk realtime setup with mysql reading the extensions table. very slick. I don't see a way to insert "include => context" into the extensions table. I don't want to have to edit the extensions.conf file each time I want to add a new context. Can you shed any light?
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