IRC log for #asterisk on 20210202

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01:01.30sawgoodI got an Asterisk 16.16.0 box (chan_SIP) port 5060: which can register to another Asterisk 11.x.x box, but not to an Asterisk 16.9 box
01:02.39sawgoodthe Asterisk 16.9 (Box C): sends back an unauthorized message (and) I know the sip.conf entries on both sides match and are correct
01:03.35sawgoodBox A to Box B (Works) but not Box A to Box C (unauthorized)
01:10.25SamotIs it supposed to challenge for auth?
01:12.21sawgoodhey Samot ...
01:13.27sawgoodBox C (Asterisk 16.9) has other chan_SIP boxes registered (OK) to it
01:13.43sawgoodI'm not sure where in sip.conf that would go (for box C)
01:14.28sawgoodI don't how to tell box C to challenge (because) the other accounts registered to it are Asterisk 11 boxes
01:15.26SamotIs Box A registered to Box C?
01:16.13sawgoodBox A is reigstered to Box B (Asterisk 11) .... Box A will not register to Box C (Asterisk 16.9)
01:16.36sawgoodBox A = Asterisk 16.16
01:17.17sawgoodBox C sends back a 401-unauthroized to Box A
01:17.55SamotOK that's a normal step.
01:18.07SamotThe 401 Unauthorized is part of the auth process.
01:18.24SamotIs Box A sending a new REGISTER with the updated auth details?
01:19.09sawgoodyes: in fact, to test: I changed the username to make sure it is coming over ...
01:19.30SamotAnd what happens with the new REGISTER is sent with the auth digest details?
01:19.37sawgoodBox A: has the regiser => statement in sip.conf
01:19.49SamotOK.
01:19.58sawgoodthe same thing: 401-unauthorized
01:20.37SamotAre you sure it's a *new* REGISTER? It has the www/auth digest details it?
01:20.49SamotIt's just not re-transmitting the previous one?
01:21.36sawgoodI'm not sure how to force a new one?
01:21.48sawgoodI did: sip reload
01:21.54SamotThe 401 Unauthorized is how that is forced.
01:22.02SamotYou understand the flow of the process, right?
01:22.20sawgoodyes and to help I have sngrep running on both Box A and Box C
01:22.24SamotOK
01:22.42SamotSo you see REGISTER -> then a 401 reply back to Box A
01:22.53sawgoodI was thinking: maybe Asterisk 16 registers differntly than Asterisk 11 for chan_SIP?
01:22.55SamotIs box A sending a *new* REGISTER with www-digest details in it?
01:22.57SamotNo.
01:22.58SamotJFC.
01:23.02SamotAnswer the question.
01:23.18sawgoodyes, in sngrep the registration is repeating ... over and over
01:23.25SamotFFS
01:23.26sawgoodsame register and then unauthorized
01:23.32SamotIs it a re-transmission of the same request?
01:23.38sawgoodprobably
01:23.41SamotOr is it a *new* with auth-digest?
01:24.02sawgoodI'd like to answer you, but you are not really making me understand what to look for
01:24.12SamotI've said it numerous tims.
01:24.14Samottimes
01:24.21SamotThis is the flow of the auth process.
01:24.23sawgoodwell: you might have, but I'm not getting it
01:24.36SamotOK so you do not understand the auth process.
01:24.39sawgoodI can stop Asterisk on both sides and restart them to be sure it is fresh?
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01:24.56SamotThe client is going to send a REGISTER request, there's no auth details in the request.
01:25.04sawgoodagreed
01:25.05SamotThe server is going to send back a 401 challenge.
01:25.17SamotThe client then sends back a *new* REGISTER with auth details in it.
01:25.19sawgoodI don't see the challenge: just the 401-unauthorized
01:25.26SamotThe 401 IS THE CHALLENGE
01:25.29sawgoodoh ok, Samot: that process ... right
01:25.45sawgoodok so only the registration and the unauthorized are happening
01:26.13Samotwow.
01:26.15SamotMan..
01:26.18sawgoodBox C: might need something to challenge then ...
01:26.29SamotIf Box A keeps re-transmitting the same REGISTER over and over again
01:26.30sawgoodbecuase it works from Box A to Box B
01:26.34SamotIt's not getting the 401 reply
01:26.37sawgoodyes it does, Samot
01:26.45SamotSo where are you running sngrep?
01:26.48SamotWhich side?
01:26.51sawgoodon both boxes
01:26.57sawgoodBox A and Box C
01:27.04SamotShow me one of these failed attempts
01:27.06sawgoodThanks for your tips
01:27.47SamotFrom Box A, show me a failed REGISTER attempt that keeps repeating
01:28.01sawgoodI'm going to look at the flow from Box A to Box B: (that works) and see if that flow looks different
01:28.06sawgoodjust give me a min
01:29.16sawgoodah now I see more:  Box A sends to Box B (then) unauthorized comes back ... then REG happens again: and then 200 OK happens
01:29.27sawgoodBox A to Box B is registred
01:29.39sawgoodBox A to Box C = never does that 200 OK ...
01:30.18SamotAnd you haven't confirmed if the updated register is happening.
01:30.31sawgoodthe "authorization digest" comes in that 2nd register attempt
01:30.41SamotOn Box C?
01:30.50sawgoodon box A
01:30.58sawgoodfor Box A to Box B
01:31.02SamotNo.
01:31.08SamotI'm asking about Box C
01:31.29sawgoodright: but I stopped and watch the flow on Box A to Box B (in sngrep)
01:31.42sawgoodto see the reg process complete ... to compare to Box A to Box C
01:33.48SamotWell something between Box A and Box C is misconfigured.
01:34.00sawgoodyeah: Samot is might be ...
01:35.09sawgoodBox C: has other boxes registered to it (But) they are Asterisk 11 ...
01:35.20sawgoodBox C is treating Box A differently ...
01:36.50SamotHow so?
01:37.07SamotIt is challenging the request
01:37.27SamotThat is what it should do
01:37.29sawgoodBox A sends the REG to Box C (but) only gets the 401-back (the) 2nd authroization with the digest never happens
01:38.07sawgoodmabye Box C needs to challenge Box A (For its username/secret)?
01:38.18SamotAnd that is due to something on Box A's side
01:38.24sawgoodreally?
01:38.25SamotIt is
01:38.33SamotIt sends a 401 reply
01:38.41sawgoodyes it does!
01:38.42SamotThat is the challenge
01:39.08SamotBox A is either not getting it or responding to it.
01:39.19SamotDont know cant see a debug from Box A
01:39.39sawgoodBox A to Box B (works) and the same config in sip.conf for Box C is used (Short the HOST= part
01:40.39Kobazfun
01:40.50SamotShow me a debug from Box A for the failure
01:40.53Kobazgetting a lot of  acl.c:833 resolve_first: Unable to lookup '38.130.255[D.68'    these days
01:41.15sawgoodok will do
01:47.33sawgoodgot it, Samot: thanks to your tips Box A now registers to Box C (and)
01:47.50sawgoodit was becasuse that send packet with the (nonce) for auth was not coming back ...
01:48.12SamotRight
01:48.20sawgoodthat 2nd packet with the (nonce) statment for authorization was not making it back to Box A ...
01:48.31sawgoodwonderful, your help is so appreciated!
01:48.44sawgoodI got the 200 (OK) now!
01:49.21sawgoodFor some reason: even with port 5060 open: something was not making it back into Box A from Box C ... (don't know)
01:49.27Kobazhnmmmmm
01:49.40KobazI don't see any options in CHANNEL to get the current codec
01:51.36Kobazset_format: Unable to find a codec translation path: (gsm|slin48|ulaw|slin) -> (opus)
01:51.44Kobazwhat native format can convert to opus?
01:51.54Kobazwell, not even native... just a format in general?
01:52.30Kobazooooh maybe codec_resample is needed
01:52.50KobazModule 'codec_resample' already loaded and running. doh
01:59.29Kobazooooooh: codec_opus.so                  OPUS Coder/Decoder                       0          Not Running      extended
01:59.31Kobazthat might do it
02:05.37sawgoodSamot: thanks again!
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05:42.36ckbhey friends! if I have a pjsip, how can I tell decide which device has the active call?
05:43.11ckbsorry, was on the phone... I'm trying to differentiate what channels are active on what device.
05:54.54ckbI'm thinking I have to use the bridge ID?
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09:51.57FaizanNMorning everyone. I wanted to know how I can precache audio files in Asterisk before creating a call/Dial or using playback.
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10:54.35ruiedhello. Can I set max_reties to infinite? in pjsip.conf ? When Internet fails asterisk seems stopping the attempt to make outbound registation
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17:54.34ruiedHello. I am receiving calls from an Internet provider getting to a local extension at a grandstream phone. When forward the incoming calls to an externa mobile number, I have no audio. At first the provider was blocking the calls transfered with Callerid(rdnis)=<somethin> and I seted it to '' . Now I can establish the forwarding call but no audio... what can it be?
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18:04.22TandyUKruied: theres a fairly long list, you should speak to your providers tech support
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18:04.35TandyUKdepending how you do it, you can either....
18:04.37nbjoergruied: check that SRTP is *not* used
18:04.55nbjoergthat was one of the show stoppers I hit with Grandstream in the past
18:04.56TandyUKcall comes into provider >> sent to phone >> phone dials a new call out >> mobile
18:04.57TandyUKOR
18:05.04nbjoergbut there are other options
18:05.12TandyUKcall comes into provider >>  in their system, goes directly to your mobile
18:05.38TandyUKif the latter, not sure what would cause audio issues, but if its the former, theres another load of things that coul;d be wrong on the extension
18:06.09TandyUKwe would call the latter option "night mode", but it may also have other names
18:44.51*** join/#asterisk catphish (~charlie@unaffiliated/catphish)
18:47.29catphishbit of an noob architecture question: in an asterisk-as-a-service environment, is it usual to provision separate instances of asterisk for the edge (to handle communication with POTS peets, and generate billing CDRs) and the media server (to handle end end user devices and more complex tasks)?
18:48.15catphishin the past i've deployed a single server, and been bitten by the complexity of processing CDRs when people are doing things like transferring calls
18:48.36catphishor are there just a million ways to do this, none of which are technically wrong?
18:50.43SamotAsterisk isn't really a switch
18:51.04SamotAnd its CDRs can be quite messy
18:51.17catphishno, i suppose it's designed more as a media server, but i have been abusing it as a switch for the entirity of my career
18:51.32catphishwhich i suppose it why i'm in the position of wondering if i need something separate to handle billing
18:53.13catphishi'm very much hoping that if i place an asterisk at the edge (between the PBX and the external peers), its CDRs will be accurate and simple, as opposed to the messy ones generated by a feature-rich PBX
18:53.40catphishbut i wondered if this was common, or if i was missing something obviously better
18:54.04SamotThe CDRs in Asterisk are messy
18:54.18SamotA single call can have multiple records.
18:54.39SamotIf you want to have something between the PSTN and do billing you should be using a switch/proxy.
18:54.42SamotKamailio does wonders.
18:55.23catphishkamailio might work well, i've not used it to generate CDRs, but i'll look into that
18:56.03catphishwill asterisk not generate simple one-cdr-per-call CDRs in that scenario then?
18:56.31catphishor is this one of those "it really depends" things?
19:00.48SamotNo, it does not.
19:02.51catphishwell that's news to me :(
19:03.19catphishi'll definitely investigate that more thoroughly
19:04.25SamotA single CDR only tracks information about a single path of communication between two endpoints. In many scenarios, there will be multiple paths of communication between multiple parties, even in a single "call". Each path of communication results in a new CDR, each representing the communication between two endpoints
19:04.34Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification
19:07.07SamotBeen like that for almost a decade.
19:07.51catphishsure, but what i need to understand is whether in in my scenario, a "billable call", and a "single path of communication between two endpoints" are synonymous
19:08.52catphishclearly i need to do some research here, both in terms of how asterisk and kamailio handle this
19:09.50SamotKamailio handles it as 1 call = 1 record
19:10.29SamotUnless you are doing multi-leg CDRs for forwarding, etc.
19:13.33SamotEven more so, if you're are doing failover in case the primary carrier is down Kamailio will still treat that as a single call versus Asterisk that will treat that as two calls.
19:14.05SamotBecause for Asterisk that is exact what it will be. Two calls.
19:15.40Samotcatphish: What type of scenario is this overall? A solution for a single PBX or a solution for multiple PBXes?
19:19.10catphishthis is something that will start simple and grow steadily, initially it will be a single PBX hosting a small number of separate users with trivial requirements (a phone, a number, voicemail), as it scales, i will likely add additional PBXs with additional features, so i'm looking to preemptively optimize in terms of getting a switch in place at the edge to handle billing and routing
19:19.56catphishpreviously, i've just installed asterisk, started piling users and features onto it, and suffered later :)
19:20.16SamotSo this is going to be providing voice services en mass
19:20.31catphishyes
19:20.42SamotUse Kamailio
19:21.26catphishperhaps the correct solution is to start with kamailio, and add asterisk as i need the PBX functionality
19:21.31catphish*asterisks
19:21.43SamotYou need it for voicemail
19:22.00SamotOr doing any DTMF processing
19:22.36catphishi assumed i would for voicemail, where would i put the UA registration?
19:23.03catphishmy instinct is to trust most things to asterisk and just use kamailio for billing and network edge routing
19:23.03SamotWell I do it at Kamailio
19:23.41catphishi suppose kamalio can do everything, and only unanswered calls need to reach asterisk
19:23.42SamotI use it on both sides
19:23.53SamotIt is a proxy
19:24.08SamotIt doesnt do media
19:24.56catphishah okay, i'll almost definitely want to proxy media, so will look at using asterisk for all media, and kamailion on one or both sides
19:25.30SamotThis a new venture?
19:26.10catphishnew employment, coming into a small-ish FTTH ISP and looking to add voice services
19:26.35SamotI cant remember, you in the US?
19:26.47catphishUK
19:27.00catphishi previously worked on https://www.dial9.co.uk/
19:27.06SamotWell then you need to be aware of regulations
19:27.30SamotLike GDPR has rules about staring personal data
19:27.55SamotHaving callerid name with the number in the record is enough to give away personal data
19:28.03catphishi'm well aware of the UK data protection and (most of) the telecoms regulation
19:28.08SamotOk
19:28.49catphishcallerid specifications are a whole fun mess of their own
19:29.14catphishbut thanks for the reminder
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