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| 01:01.30 | sawgood | I 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 |
| 01:02.39 | sawgood | the Asterisk 16.9 (Box C): sends back an unauthorized message (and) I know the sip.conf entries on both sides match and are correct |
| 01:03.35 | sawgood | Box A to Box B (Works) but not Box A to Box C (unauthorized) |
| 01:10.25 | Samot | Is it supposed to challenge for auth? |
| 01:12.21 | sawgood | hey Samot ... |
| 01:13.27 | sawgood | Box C (Asterisk 16.9) has other chan_SIP boxes registered (OK) to it |
| 01:13.43 | sawgood | I'm not sure where in sip.conf that would go (for box C) |
| 01:14.28 | sawgood | I don't how to tell box C to challenge (because) the other accounts registered to it are Asterisk 11 boxes |
| 01:15.26 | Samot | Is Box A registered to Box C? |
| 01:16.13 | sawgood | Box A is reigstered to Box B (Asterisk 11) .... Box A will not register to Box C (Asterisk 16.9) |
| 01:16.36 | sawgood | Box A = Asterisk 16.16 |
| 01:17.17 | sawgood | Box C sends back a 401-unauthroized to Box A |
| 01:17.55 | Samot | OK that's a normal step. |
| 01:18.07 | Samot | The 401 Unauthorized is part of the auth process. |
| 01:18.24 | Samot | Is Box A sending a new REGISTER with the updated auth details? |
| 01:19.09 | sawgood | yes: in fact, to test: I changed the username to make sure it is coming over ... |
| 01:19.30 | Samot | And what happens with the new REGISTER is sent with the auth digest details? |
| 01:19.37 | sawgood | Box A: has the regiser => statement in sip.conf |
| 01:19.49 | Samot | OK. |
| 01:19.58 | sawgood | the same thing: 401-unauthorized |
| 01:20.37 | Samot | Are you sure it's a *new* REGISTER? It has the www/auth digest details it? |
| 01:20.49 | Samot | It's just not re-transmitting the previous one? |
| 01:21.36 | sawgood | I'm not sure how to force a new one? |
| 01:21.48 | sawgood | I did: sip reload |
| 01:21.54 | Samot | The 401 Unauthorized is how that is forced. |
| 01:22.02 | Samot | You understand the flow of the process, right? |
| 01:22.20 | sawgood | yes and to help I have sngrep running on both Box A and Box C |
| 01:22.24 | Samot | OK |
| 01:22.42 | Samot | So you see REGISTER -> then a 401 reply back to Box A |
| 01:22.53 | sawgood | I was thinking: maybe Asterisk 16 registers differntly than Asterisk 11 for chan_SIP? |
| 01:22.55 | Samot | Is box A sending a *new* REGISTER with www-digest details in it? |
| 01:22.57 | Samot | No. |
| 01:22.58 | Samot | JFC. |
| 01:23.02 | Samot | Answer the question. |
| 01:23.18 | sawgood | yes, in sngrep the registration is repeating ... over and over |
| 01:23.25 | Samot | FFS |
| 01:23.26 | sawgood | same register and then unauthorized |
| 01:23.32 | Samot | Is it a re-transmission of the same request? |
| 01:23.38 | sawgood | probably |
| 01:23.41 | Samot | Or is it a *new* with auth-digest? |
| 01:24.02 | sawgood | I'd like to answer you, but you are not really making me understand what to look for |
| 01:24.12 | Samot | I've said it numerous tims. |
| 01:24.14 | Samot | times |
| 01:24.21 | Samot | This is the flow of the auth process. |
| 01:24.23 | sawgood | well: you might have, but I'm not getting it |
| 01:24.36 | Samot | OK so you do not understand the auth process. |
| 01:24.39 | sawgood | I can stop Asterisk on both sides and restart them to be sure it is fresh? |
| 01:24.51 | *** join/#asterisk Janos (~textual@201.204.94.76) |
| 01:24.56 | Samot | The client is going to send a REGISTER request, there's no auth details in the request. |
| 01:25.04 | sawgood | agreed |
| 01:25.05 | Samot | The server is going to send back a 401 challenge. |
| 01:25.17 | Samot | The client then sends back a *new* REGISTER with auth details in it. |
| 01:25.19 | sawgood | I don't see the challenge: just the 401-unauthorized |
| 01:25.26 | Samot | The 401 IS THE CHALLENGE |
| 01:25.29 | sawgood | oh ok, Samot: that process ... right |
| 01:25.45 | sawgood | ok so only the registration and the unauthorized are happening |
| 01:26.13 | Samot | wow. |
| 01:26.15 | Samot | Man.. |
| 01:26.18 | sawgood | Box C: might need something to challenge then ... |
| 01:26.29 | Samot | If Box A keeps re-transmitting the same REGISTER over and over again |
| 01:26.30 | sawgood | becuase it works from Box A to Box B |
| 01:26.34 | Samot | It's not getting the 401 reply |
| 01:26.37 | sawgood | yes it does, Samot |
| 01:26.45 | Samot | So where are you running sngrep? |
| 01:26.48 | Samot | Which side? |
| 01:26.51 | sawgood | on both boxes |
| 01:26.57 | sawgood | Box A and Box C |
| 01:27.04 | Samot | Show me one of these failed attempts |
| 01:27.06 | sawgood | Thanks for your tips |
| 01:27.47 | Samot | From Box A, show me a failed REGISTER attempt that keeps repeating |
| 01:28.01 | sawgood | I'm going to look at the flow from Box A to Box B: (that works) and see if that flow looks different |
| 01:28.06 | sawgood | just give me a min |
| 01:29.16 | sawgood | ah now I see more: Box A sends to Box B (then) unauthorized comes back ... then REG happens again: and then 200 OK happens |
| 01:29.27 | sawgood | Box A to Box B is registred |
| 01:29.39 | sawgood | Box A to Box C = never does that 200 OK ... |
| 01:30.18 | Samot | And you haven't confirmed if the updated register is happening. |
| 01:30.31 | sawgood | the "authorization digest" comes in that 2nd register attempt |
| 01:30.41 | Samot | On Box C? |
| 01:30.50 | sawgood | on box A |
| 01:30.58 | sawgood | for Box A to Box B |
| 01:31.02 | Samot | No. |
| 01:31.08 | Samot | I'm asking about Box C |
| 01:31.29 | sawgood | right: but I stopped and watch the flow on Box A to Box B (in sngrep) |
| 01:31.42 | sawgood | to see the reg process complete ... to compare to Box A to Box C |
| 01:33.48 | Samot | Well something between Box A and Box C is misconfigured. |
| 01:34.00 | sawgood | yeah: Samot is might be ... |
| 01:35.09 | sawgood | Box C: has other boxes registered to it (But) they are Asterisk 11 ... |
| 01:35.20 | sawgood | Box C is treating Box A differently ... |
| 01:36.50 | Samot | How so? |
| 01:37.07 | Samot | It is challenging the request |
| 01:37.27 | Samot | That is what it should do |
| 01:37.29 | sawgood | Box A sends the REG to Box C (but) only gets the 401-back (the) 2nd authroization with the digest never happens |
| 01:38.07 | sawgood | mabye Box C needs to challenge Box A (For its username/secret)? |
| 01:38.18 | Samot | And that is due to something on Box A's side |
| 01:38.24 | sawgood | really? |
| 01:38.25 | Samot | It is |
| 01:38.33 | Samot | It sends a 401 reply |
| 01:38.41 | sawgood | yes it does! |
| 01:38.42 | Samot | That is the challenge |
| 01:39.08 | Samot | Box A is either not getting it or responding to it. |
| 01:39.19 | Samot | Dont know cant see a debug from Box A |
| 01:39.39 | sawgood | Box A to Box B (works) and the same config in sip.conf for Box C is used (Short the HOST= part |
| 01:40.39 | Kobaz | fun |
| 01:40.50 | Samot | Show me a debug from Box A for the failure |
| 01:40.53 | Kobaz | getting a lot of acl.c:833 resolve_first: Unable to lookup '38.130.255[D.68' these days |
| 01:41.15 | sawgood | ok will do |
| 01:47.33 | sawgood | got it, Samot: thanks to your tips Box A now registers to Box C (and) |
| 01:47.50 | sawgood | it was becasuse that send packet with the (nonce) for auth was not coming back ... |
| 01:48.12 | Samot | Right |
| 01:48.20 | sawgood | that 2nd packet with the (nonce) statment for authorization was not making it back to Box A ... |
| 01:48.31 | sawgood | wonderful, your help is so appreciated! |
| 01:48.44 | sawgood | I got the 200 (OK) now! |
| 01:49.21 | sawgood | For some reason: even with port 5060 open: something was not making it back into Box A from Box C ... (don't know) |
| 01:49.27 | Kobaz | hnmmmmm |
| 01:49.40 | Kobaz | I don't see any options in CHANNEL to get the current codec |
| 01:51.36 | Kobaz | set_format: Unable to find a codec translation path: (gsm|slin48|ulaw|slin) -> (opus) |
| 01:51.44 | Kobaz | what native format can convert to opus? |
| 01:51.54 | Kobaz | well, not even native... just a format in general? |
| 01:52.30 | Kobaz | ooooh maybe codec_resample is needed |
| 01:52.50 | Kobaz | Module 'codec_resample' already loaded and running. doh |
| 01:59.29 | Kobaz | ooooooh: codec_opus.so OPUS Coder/Decoder 0 Not Running extended |
| 01:59.31 | Kobaz | that might do it |
| 02:05.37 | sawgood | Samot: thanks again! |
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| 05:42.36 | ckb | hey friends! if I have a pjsip, how can I tell decide which device has the active call? |
| 05:43.11 | ckb | sorry, was on the phone... I'm trying to differentiate what channels are active on what device. |
| 05:54.54 | ckb | I'm thinking I have to use the bridge ID? |
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| 09:51.57 | FaizanN | Morning 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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| 10:54.35 | ruied | hello. 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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| 17:54.34 | ruied | Hello. 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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| 18:04.22 | TandyUK | ruied: theres a fairly long list, you should speak to your providers tech support |
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| 18:04.35 | TandyUK | depending how you do it, you can either.... |
| 18:04.37 | nbjoerg | ruied: check that SRTP is *not* used |
| 18:04.55 | nbjoerg | that was one of the show stoppers I hit with Grandstream in the past |
| 18:04.56 | TandyUK | call comes into provider >> sent to phone >> phone dials a new call out >> mobile |
| 18:04.57 | TandyUK | OR |
| 18:05.04 | nbjoerg | but there are other options |
| 18:05.12 | TandyUK | call comes into provider >> in their system, goes directly to your mobile |
| 18:05.38 | TandyUK | if 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 |
| 18:06.09 | TandyUK | we would call the latter option "night mode", but it may also have other names |
| 18:44.51 | *** join/#asterisk catphish (~charlie@unaffiliated/catphish) |
| 18:47.29 | catphish | bit 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)? |
| 18:48.15 | catphish | in 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 |
| 18:48.36 | catphish | or are there just a million ways to do this, none of which are technically wrong? |
| 18:50.43 | Samot | Asterisk isn't really a switch |
| 18:51.04 | Samot | And its CDRs can be quite messy |
| 18:51.17 | catphish | no, 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 |
| 18:51.32 | catphish | which i suppose it why i'm in the position of wondering if i need something separate to handle billing |
| 18:53.13 | catphish | i'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 |
| 18:53.40 | catphish | but i wondered if this was common, or if i was missing something obviously better |
| 18:54.04 | Samot | The CDRs in Asterisk are messy |
| 18:54.18 | Samot | A single call can have multiple records. |
| 18:54.39 | Samot | If you want to have something between the PSTN and do billing you should be using a switch/proxy. |
| 18:54.42 | Samot | Kamailio does wonders. |
| 18:55.23 | catphish | kamailio might work well, i've not used it to generate CDRs, but i'll look into that |
| 18:56.03 | catphish | will asterisk not generate simple one-cdr-per-call CDRs in that scenario then? |
| 18:56.31 | catphish | or is this one of those "it really depends" things? |
| 19:00.48 | Samot | No, it does not. |
| 19:02.51 | catphish | well that's news to me :( |
| 19:03.19 | catphish | i'll definitely investigate that more thoroughly |
| 19:04.25 | Samot | A 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 |
| 19:04.34 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
| 19:07.07 | Samot | Been like that for almost a decade. |
| 19:07.51 | catphish | sure, 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 |
| 19:08.52 | catphish | clearly i need to do some research here, both in terms of how asterisk and kamailio handle this |
| 19:09.50 | Samot | Kamailio handles it as 1 call = 1 record |
| 19:10.29 | Samot | Unless you are doing multi-leg CDRs for forwarding, etc. |
| 19:13.33 | Samot | Even 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. |
| 19:14.05 | Samot | Because for Asterisk that is exact what it will be. Two calls. |
| 19:15.40 | Samot | catphish: What type of scenario is this overall? A solution for a single PBX or a solution for multiple PBXes? |
| 19:19.10 | catphish | this 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 |
| 19:19.56 | catphish | previously, i've just installed asterisk, started piling users and features onto it, and suffered later :) |
| 19:20.16 | Samot | So this is going to be providing voice services en mass |
| 19:20.31 | catphish | yes |
| 19:20.42 | Samot | Use Kamailio |
| 19:21.26 | catphish | perhaps the correct solution is to start with kamailio, and add asterisk as i need the PBX functionality |
| 19:21.31 | catphish | *asterisks |
| 19:21.43 | Samot | You need it for voicemail |
| 19:22.00 | Samot | Or doing any DTMF processing |
| 19:22.36 | catphish | i assumed i would for voicemail, where would i put the UA registration? |
| 19:23.03 | catphish | my instinct is to trust most things to asterisk and just use kamailio for billing and network edge routing |
| 19:23.03 | Samot | Well I do it at Kamailio |
| 19:23.41 | catphish | i suppose kamalio can do everything, and only unanswered calls need to reach asterisk |
| 19:23.42 | Samot | I use it on both sides |
| 19:23.53 | Samot | It is a proxy |
| 19:24.08 | Samot | It doesnt do media |
| 19:24.56 | catphish | ah 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 |
| 19:25.30 | Samot | This a new venture? |
| 19:26.10 | catphish | new employment, coming into a small-ish FTTH ISP and looking to add voice services |
| 19:26.35 | Samot | I cant remember, you in the US? |
| 19:26.47 | catphish | UK |
| 19:27.00 | catphish | i previously worked on https://www.dial9.co.uk/ |
| 19:27.06 | Samot | Well then you need to be aware of regulations |
| 19:27.30 | Samot | Like GDPR has rules about staring personal data |
| 19:27.55 | Samot | Having callerid name with the number in the record is enough to give away personal data |
| 19:28.03 | catphish | i'm well aware of the UK data protection and (most of) the telecoms regulation |
| 19:28.08 | Samot | Ok |
| 19:28.49 | catphish | callerid specifications are a whole fun mess of their own |
| 19:29.14 | catphish | but thanks for the reminder |
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