| 00:14.53 | *** join/#asterisk Katty (uid62315@gateway/web/irccloud.com/x-xwmqjcstapovxvmb) |
| 00:19.46 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:19.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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 |
| 00:34.35 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 00:35.08 | sawgood | Question: is there way to set a [context] in sip.conf (so that) when SIP calls come in (the) FROM field is not used as part of the authentication? example: Let anything be in the FROM field (instead) of requiring the "accountcode="? |
| 00:36.12 | Samot | That would require the incoming SIP request to have a header for those. |
| 00:36.25 | Samot | "accountcode" is not a standard SIP header |
| 00:38.57 | sawgood | problem is: Cisco Call Manager is sending (different)"phone numbers" in the FROM field at times (and) most of the time they do not match the context or accountcode (and) Asterisk is sending them back a "404" |
| 00:39.15 | Samot | Uhm. |
| 00:39.18 | Samot | So? |
| 00:39.21 | drmessano | lol |
| 00:39.25 | Samot | They are incoming calls |
| 00:39.32 | sawgood | outgoing, sir |
| 00:39.51 | Samot | Not if the Cisco CM is sending them to the PBX |
| 00:40.06 | Samot | Or do you have the Cisco CM as an "endpoint"? |
| 00:40.26 | sawgood | Call Manager sends to an Asterisk SBC (for) outbound calls: and inside of Call Manager (the customer) has changed their "diversion header" and or something else (and) thus calls are not passing through the Asterisk SBC |
| 00:40.37 | sawgood | Cisco CM = customer site (their PBX) |
| 00:40.43 | Samot | First off |
| 00:40.47 | Samot | Asterisk != SBC |
| 00:40.50 | Samot | At all. |
| 00:40.51 | sawgood | <PROTECTED> |
| 00:41.14 | Samot | FROM is how Auth is done. |
| 00:41.16 | sawgood | Asterisk = B2BUA mode |
| 00:41.31 | Samot | Asterisk is not a SBC. |
| 00:41.36 | Samot | It is a B2BUA |
| 00:41.39 | drmessano | Thats it's only mode |
| 00:41.43 | sawgood | excellent .... |
| 00:41.43 | Samot | There is a difference. |
| 00:41.52 | sawgood | there is ... for sure ... |
| 00:42.02 | drmessano | What is B2BUA mode? |
| 00:42.05 | Samot | Then why are you referring to it as a SBC |
| 00:42.09 | Samot | In B2BUA mode? |
| 00:42.14 | sawgood | because it is just cool to call it that! |
| 00:42.20 | Samot | Oh dear god. |
| 00:42.36 | sawgood | SBC in B2BUA mode running Asterisk |
| 00:42.37 | sawgood | ha! |
| 00:42.39 | drmessano | SBC isn't a role |
| 00:42.46 | drmessano | It's a specific device |
| 00:42.51 | Samot | Try to sound like you have some knowledge of what you are doing. |
| 00:43.12 | sawgood | oh ok ... if you say so ... |
| 00:43.18 | Samot | You're asking two different questions in two channels and in both case the concepts are not understood. |
| 00:43.42 | sawgood | I have different concerns (one is raw asterisk) and one is FPBX using Asterisk as a PBX |
| 00:43.50 | sawgood | asterisk is also a PBX not just a B2BUA |
| 00:43.58 | Samot | Uhm. |
| 00:43.59 | Samot | Dude. |
| 00:44.01 | drmessano | LOL |
| 00:44.03 | sawgood | but we can leave that alone for now ... no need to nit-pick |
| 00:44.04 | Samot | By default a PBX is a B2BUA |
| 00:44.10 | drmessano | Asterisk is ALWAYS a B2BUA |
| 00:44.15 | drmessano | Asterisk is SOMETIMES a PBX |
| 00:44.17 | drmessano | Asterisk is ALWAYS a B2BUA |
| 00:44.22 | sawgood | no it is not, but I'll let say that |
| 00:44.26 | sawgood | right that is combo, drmessano |
| 00:44.30 | sawgood | cool one! |
| 00:45.04 | sawgood | so, as a B2BUA, Cisco Call Manager is now sending me various things in the FROM field (and) not what they were sending for many years |
| 00:45.22 | Samot | Then tell them to fix their shit. |
| 00:45.24 | Samot | You are the provider. |
| 00:45.24 | sawgood | I think changes in CM were made (and) the SIP invite has different FROM fields for calls |
| 00:45.31 | Samot | They do things how you tell them. |
| 00:45.36 | Samot | Not the other way around. |
| 00:45.43 | sawgood | Samot: you're the best, man! |
| 00:46.14 | sawgood | Asterisk can also be a media-gateway (in with SIP out with PRI) for example |
| 00:46.54 | drmessano | But it's still a B2BUA |
| 00:47.17 | sawgood | mabye it is a B2BUA with a "role" as a media-gateway |
| 00:47.23 | sawgood | is that ok to say? |
| 00:47.44 | drmessano | No, you actually don't need to say half of that |
| 00:48.31 | sawgood | I remember years back agruing B2BUA role with Patton and what not |
| 00:48.52 | sawgood | they had just released a true SBC appliance and I couldn't get it to work easliy in front of Asterisk |
| 00:54.55 | sawgood | So, when Call Manager sends the outbound call to the Asterisk B2BUA (what) is used in the [context] for Cisco Call Manger (to check) the FROM field with? Is it the [context] name (and/or) the entry in the context labelled username (please) forgot I mentioned accountcode= (my bad) |
| 00:55.54 | sawgood | in other words: does the [context] and username= need to match (and) what if they are differnt (which) is used against the FROM field (SIP invite)? |
| 00:56.02 | Samot | No. |
| 00:56.09 | Samot | The context is NOT used for AUTH |
| 00:56.26 | Samot | User, IP, Port |
| 00:56.33 | Samot | Those are the three things used. |
| 00:56.35 | sawgood | right-on ... a nice answer from you ... I feel like you are turning a corner |
| 00:56.45 | sawgood | username, IP, port |
| 00:56.54 | Samot | Are you trolling us? |
| 00:56.57 | Samot | Seriously? |
| 00:57.12 | sawgood | no I have another thing to run by you if you don't mind |
| 00:57.39 | sawgood | another deal ... (give me) 2 min to pull up my (NoOp) statement and run that by you ... |
| 00:58.29 | sawgood | By the way: yesterday I was stuck in the field and could not get to my GUI front end for VoIP Monitor (and) look for the SIP invite (and) I had to use sip set debug IP to watch live |
| 00:58.39 | sawgood | do you use VoIP Monitor or Homer? |
| 01:00.23 | Samot | No. |
| 01:00.28 | Samot | Niether. |
| 01:00.47 | drmessano | I use nmap |
| 01:03.05 | sawgood | hard core, but how do you log SIP invites (in a database) to exaine later? |
| 01:04.54 | sawgood | What about MOS? What do you use (VoIP Monitor) does this nicely |
| 01:07.18 | sawgood | dumber question: do you think there is a way to have: sip set debug IP (not come out on the screen) but rather dump out to a syslog server (freeing) the screen up for showing only non debug traffic (normal) SIP flow? |
| 01:08.04 | Samot | It's a live console command |
| 01:08.11 | Samot | It outputs to the console. |
| 01:10.41 | sawgood | As a baseline/example: I use this to show the outgoing CID on calls: exten => _X.,2,NoOp(Outgoing CID is: ${CALLERID(number)}) |
| 01:10.46 | sawgood | exten => _X.,2,NoOp(Outgoing CID is: ${CALLERID(number)}) |
| 01:11.08 | sawgood | as a pipe dream: do you think I could to this: |
| 01:11.39 | Samot | Are you setting those variables? |
| 01:12.25 | sawgood | yeah that is working now: but I wanted one that could "tell me" what RTP port is being used (as an example) |
| 01:12.51 | sawgood | here is what I mean: exten => _X.,2,NoOp(RTP port in use: $use the correct variable}) |
| 01:13.19 | iheartlinux | lol |
| 01:13.38 | sawgood | actually that is a bad example since Asterisk puts that out during a call (normally) |
| 01:13.47 | drmessano | Dude |
| 01:13.48 | iheartlinux | Samot: Are you trolling us? |
| 01:14.03 | drmessano | Have you ever actually looked at SIP Debug? |
| 01:14.11 | sawgood | let me think of a better one (because) if I tell you what I want it to to (you'll get mad at me) |
| 01:14.28 | Samot | Yes. |
| 01:14.34 | Samot | I'm trolling by helping. |
| 01:15.10 | sawgood | I have what I want to do (wrote up) in a .PDF (and) I've offered a bounty but nobody really gets it ... |
| 01:15.26 | sawgood | I've rasied the bounty to $200 I think at the last count |
| 01:15.35 | drmessano | Did you try mansplaining it? |
| 01:15.40 | drmessano | $200? |
| 01:15.42 | drmessano | ROFLMAO |
| 01:15.53 | drmessano | You're not serious, right? |
| 01:15.56 | iheartlinux | hahahaha |
| 01:16.23 | sawgood | yes, but it is for the SIP stack (on a phone) showing DNID on the display (on an incoming call) ... |
| 01:16.34 | drmessano | sawgood: Add a zero |
| 01:16.38 | drmessano | Then maybe |
| 01:17.00 | sawgood | the closest example I guess I could tell you (is) (With) FreePBX (they call it) Caller-ID Prefix |
| 01:17.24 | sawgood | caller-ID prefix is too 'weak' and there are line breaks at times, etc |
| 01:17.46 | drmessano | sawgood: Have you tried hiring someone? |
| 01:17.54 | sawgood | I'll try to tell you here: but it goes by many names: some call it DNID and others call it RDNID |
| 01:18.00 | iheartlinux | Samot: I was quoting you |
| 01:18.29 | sawgood | I think most SIP IP phones are limited to the standard SIP stack for what happens with caller ID (on) a LCD display |
| 01:19.03 | drmessano | Well, they are SIP |
| 01:19.12 | iheartlinux | lol |
| 01:19.13 | sawgood | for incoming calls (when) the phone rings (and) you see (date, time, CNAM, and the phone number) .... that is what you get (as) long as you have CNAM from your provider |
| 01:19.14 | drmessano | So I guess they're limited to SIP |
| 01:19.23 | drmessano | Fucking dumb SIP phones only doing SIP |
| 01:19.29 | drmessano | GOD DAMN THEM ALL |
| 01:19.36 | iheartlinux | ROFL |
| 01:19.45 | Samot | Those are just SIP headers. |
| 01:19.48 | Samot | That they set. |
| 01:19.49 | drmessano | They need to support RFC69FFFFFFFFFFFFFFFF |
| 01:20.14 | sawgood | the SIP stack does not show DNID or RDNID (open source SIP) but other makers of PBXs have rebuild their own SIP process (and) thus they do cool things with inbound calls (for example) Allworx and Panasonic for example |
| 01:20.20 | drmessano | SIP phones don't SIP hard enough |
| 01:20.39 | drmessano | Wait, what? |
| 01:20.44 | Samot | Yeah. |
| 01:22.22 | drmessano | They also don't IPv7 very well |
| 01:22.24 | sawgood | When you have an Allworx SIP phone (private SIP stack) ... and when you get an inbound call to the phone (either SIP, analog, PRI) on the LCD of the phone (along with ALL the normal CALLER ID details) you also get one more line (above them all) at the very top (which) shows the person answering the phone (which phone number was dialed to reach your phone) |
| 01:22.47 | Samot | hah. |
| 01:22.48 | Samot | OK. |
| 01:22.52 | drmessano | Dude, what |
| 01:23.01 | sawgood | long-winded ... but that is the point |
| 01:23.09 | Samot | So they are just sending caller id data to the phone |
| 01:23.14 | sawgood | getting DNID or RDNID (shown) on the pohne |
| 01:23.14 | Samot | In the SIP message |
| 01:23.15 | drmessano | ^ |
| 01:23.26 | Samot | Because it's their PBX |
| 01:23.29 | Samot | And their phones |
| 01:23.40 | Samot | So they can kinda program them to look out for custom SIP headers |
| 01:23.44 | Samot | And do stuff with them. |
| 01:23.45 | sawgood | well, I have it pulled out via Asterisk (let) me show you what I mean .... |
| 01:23.58 | drmessano | Please, put it away |
| 01:24.06 | drmessano | We don't need to see that |
| 01:24.08 | sawgood | let me show you my (noOp) statment for this (how) asterisk "pulls" out DNID for me |
| 01:24.11 | drmessano | Think of the children |
| 01:24.23 | drmessano | Oh god, now he's gonna pull out |
| 01:24.48 | drmessano | covers his SBC |
| 01:24.58 | sawgood | exten => _X.,1,NoOp("DNID is: ${CALLERID(dnid)}, Caller Number is: ${CALLERID(num)}, CNAM is: ${CALLERID(name)}") |
| 01:25.11 | Samot | OK |
| 01:25.15 | Samot | That's standard. |
| 01:25.16 | drmessano | So MessageSend that to the phone |
| 01:25.17 | sawgood | so, Asterisk can do this ... show you the phone number dialed .... to reach the PBX |
| 01:25.25 | drmessano | SIP Message |
| 01:25.29 | drmessano | Done |
| 01:25.36 | sawgood | I just want: the DNID to also make it to the SIP phone .... |
| 01:25.41 | drmessano | So MessageSend that to the phone |
| 01:25.43 | drmessano | SIP Message |
| 01:25.44 | drmessano | Done |
| 01:26.01 | sawgood | drmessano: if that works ... collect the bounty from me? |
| 01:26.15 | drmessano | I don't need it... I'm rich |
| 01:26.17 | drmessano | But go for it |
| 01:26.50 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend |
| 01:27.05 | sawgood | hmmm ... I do have a question ... |
| 01:27.15 | Samot | When do you not? |
| 01:27.40 | sawgood | is MessageSend: part of Asterisk 11 or only 13? |
| 01:27.54 | drmessano | 11 |
| 01:28.02 | drmessano | I pasted the most recent do |
| 01:28.03 | drmessano | I pasted the most recent doc |
| 01:28.10 | sawgood | reading ... |
| 01:29.07 | sawgood | I guess the MessageSend: would need to be done/run on the PBX the SIP phone (Yealink) is registered to (and) not pushed to the PBX from the B2BUA in front of the PbX |
| 02:06.38 | *** join/#asterisk iheartlinux (~jwpierce3@mail.trunkmasters.com) |
| 02:59.42 | *** join/#asterisk ruben23 (~OpenDIAL@112.211.86.145) |
| 03:06.49 | ruben23 | guys any help installed asterisk 11 and i get this when i tried to start it ---> asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 03:11.38 | ruben23 | any idea guys |
| 03:16.20 | Samot | Sounds like openssl isn't installed where Asterisk would like it to be |
| 03:16.40 | ruben23 | let me install |
| 03:19.41 | ruben23 | openssl is already the newest version. |
| 03:19.58 | ruben23 | Samot: openssl is already the newest version |
| 03:20.12 | Samot | I didn't say the version was wrong. |
| 03:20.23 | Samot | I said it was installed in a place Asterisk isn't looking |
| 03:21.06 | ruben23 | <PROTECTED> |
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| 04:01.04 | ruben23 | guys any help for this ---> installed asterisk 11 and i get this when i tried to start it ---> asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 04:01.13 | ruben23 | O:-) |
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| 04:50.26 | mrlorsungcu | ;-) |
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| 07:37.38 | wdoekes | ruben23: did you run ldconfig after install? |
| 07:38.54 | wdoekes | if you do 'ldd $(which asterisk)' you should be able to see where it expects libasteriskssl.so |
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| 14:17.42 | LunaLovegood | Is there a way to make voicemail less verbose in the CLI without reducing the verbosity of the rest? At the moment, my console fills with lines like "<PJSIP/XUPdd3Flry-00000117> Playing 'digits/1.slin'" whenever someonw dials into their mailbox. |
| 14:18.06 | LunaLovegood | s/someonw/someone/ |
| 14:18.32 | LunaLovegood | sorry, didn't know it woul do that |
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| 14:58.23 | Samot | LunaLovegood: That's standard for Asterisk when playing back recordings. |
| 14:58.39 | Samot | It does that every time it playsback |
| 14:59.50 | Samot | It's not limited to voicemail |
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| 16:16.02 | Ice_Strike | Hey |
| 16:34.01 | hdon | hi all :) OT question: a guy at a SIP termination provider told me that on the PSTN your codecs may be changed dynamically based on your utilization. is that true? i kinda feel like he might be confusing the PSTN with their own service |
| 16:37.48 | Samot | Depends. |
| 16:39.25 | Samot | 1) Termination providers use multiple carriers |
| 16:39.40 | Samot | 2) They rarely deal with the media of the calls, just the signalling |
| 16:40.03 | Samot | 3) They don't have their own "network" so to speak so all calls hit the PSTN for routing. |
| 16:40.12 | Samot | ^ Sometimes. |
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| 17:11.41 | drmessano | Thats why you pay for a decent provider |
| 17:11.51 | drmessano | Not .0015 cents per minute |
| 17:11.59 | drmessano | Someone that uses high quality routes |
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| 17:18.56 | Samot | It's why I use Asterisk to handle media for that. |
| 17:19.31 | Samot | So the end user can use a codec that *I* support and not have to worry about getting the wrong codec from the upstream |
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| 21:06.23 | sawgood | If in a context inside of sip.conf: (if I didn't want to use) username= (and) just allow "anything" to be accepted in the "FROM" field in the SIP invite (can) I do this? |
| 21:06.53 | sawgood | NOTE: I have IP auth set in the same context (but) I wanted to drop the username= part (and) allow anything to be in the FRON field as accepted |
| 21:22.01 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 21:31.11 | sawgood | if using IP auth (in a context) ... is username= still required? |
| 21:31.35 | sawgood | I ask because I have a need to allow "anything" in the SIP invite (FRON) to go though with outbound call attempts |
| 21:31.39 | Samot | Contexts do not auth. |
| 21:31.43 | Samot | Period. |
| 21:31.55 | Samot | Contexts are where the calls go _after_ auth |
| 21:32.53 | sawgood | what are these two statements for then |
| 21:32.57 | sawgood | deny=0.0.0.0/0.0.0.0 |
| 21:32.57 | sawgood | permit=123.13.14.56/255.255.255.255 |
| 21:33.16 | sawgood | IP auth! |
| 21:34.06 | *** join/#asterisk spicyramen (~Adium@104.132.1.77) |
| 21:34.46 | Samot | No. |
| 21:35.04 | Samot | Those are to allow or deny IPs to access it.. |
| 21:35.07 | Samot | Nothing to do with auth |
| 21:35.38 | Samot | insecure= |
| 21:36.00 | Samot | insecure=invite,port <-- means don't auth on and/or with those things. |
| 21:36.05 | Samot | Don't auth invites |
| 21:36.21 | Samot | And don't use port when validating the IP |
| 21:38.28 | sawgood | thank you! |
| 21:38.37 | Samot | 5:32:57 PM <sawgood> deny=0.0.0.0/0.0.0.0 |
| 21:38.37 | Samot | 5:32:57 PM <sawgood> permit=123.13.14.56/255.255.255.255 |
| 21:38.37 | Samot | ^^ These only come into play with the host=dynamic |
| 21:38.50 | sawgood | right ... right .... (totaly recall now) |
| 21:43.08 | sawgood | At times, Cisco Call Manager (now wants to) send "differnt" (ever changing) phone numbers in the FROM field (not jus the one they have been sending for a long time) ... they want what ever is sent in the FROM field to be accepted by Asterisk |
| 21:43.34 | sawgood | They call this their: Calling Party Transform Mask |
| 21:43.40 | Samot | OK. |
| 21:43.58 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 21:44.12 | [TK]D-Fender | <Samot> 5:32:57 PM <sawgood> permit=123.13.14.56/255.255.255.255 <- pointless |
| 21:44.16 | Samot | If they've got a static IP then just don't auth their calls. |
| 21:44.39 | [TK]D-Fender | this is effectively the same restriction as SETTING the host explicitly except that you waste your time adding 2 more lines into your config for it |
| 21:44.43 | Samot | With a user/pass..just use the IP. |
| 21:46.06 | sawgood | host= use the IP of the call manager box? |
| 21:46.30 | Samot | Yes, like any other trunk/peer you would do this with |
| 21:46.49 | sawgood | username = and secret = (remove these two) |
| 21:48.26 | sawgood | and if desired: add insecure=invite,port |
| 21:48.36 | Samot | No. |
| 21:48.40 | Samot | Required. |
| 21:48.48 | sawgood | ty! |
| 21:48.56 | Samot | They are peering to you like you peer to a provider. |
| 21:49.06 | Samot | You don't auth their incoming calls. |
| 21:49.17 | sawgood | exactly ... very sharp! |
| 21:50.57 | sawgood | one small point (which) for some reason (is escaping me) ... (dumb question) but why is host=IP (differnt) from permit=IP (if) they both are going to permit/deny on IP address? |
| 21:51.16 | Samot | host is the other end |
| 21:51.30 | Samot | If you don't know what that IP is going to be all the time... |
| 21:51.34 | Samot | host=dynamic |
| 21:51.42 | Samot | When it's dynamic, it means ANY IP is allowed |
| 21:51.51 | Samot | It auth's on the user/secret |
| 21:52.01 | sawgood | host=dynamic (also) allow SIP registration too (right)? |
| 21:52.06 | Samot | Yes. |
| 21:52.14 | Samot | That's the point |
| 21:52.26 | Samot | Most endpoints have a dynamic IP |
| 21:52.31 | sawgood | wow permit= (is there) to confirm the user/pass (only) from that one IP |
| 21:52.38 | Samot | Or a range |
| 21:52.43 | sawgood | or a range! |
| 21:52.51 | Samot | But if you know the IP |
| 21:53.04 | Samot | host=IP |
| 21:53.16 | Samot | It's not dynamic if it's always the same. |
| 21:53.20 | sawgood | what if you have a host= IP addresss (but) you STILL want the other side to "register" (for other reasons) |
| 21:53.31 | Samot | Then you don't use insecure |
| 21:53.49 | sawgood | nice! |
| 21:53.59 | Samot | Asterisk 101 |
| 21:58.02 | [TK]D-Fender | <sawgood> what if you have a host= IP addresss (but) you STILL want the other side to "register" (for other reasons) <- you can't |
| 21:58.11 | [TK]D-Fender | if you set a host you are not permitted to register |
| 21:58.26 | sawgood | re-cap question (sorry for parroting) but if you have: invite=insecure,port (which) then ignores the need to trade SIP invites for a (username,password) .... (why) then use host= with an IP address? |
| 21:58.35 | Samot | Sorry, yeah not register.. |
| 21:58.37 | [TK]D-Fender | at which point if you have to in order to function then you'd use permit/deny to restrict while being dynamic |
| 21:59.10 | sawgood | I'm stuggling a bit with the "difference" between host= and permit= |
| 21:59.24 | Samot | You're confusing AUTH with REGISTER |
| 21:59.30 | Samot | REGISTER is for INBOUND |
| 21:59.40 | Samot | Tells the other side where to SEND stuff |
| 22:00.11 | Samot | When a device "registers" it is just dropping off its location to the system. |
| 22:00.18 | sawgood | got that part ... |
| 22:00.23 | Samot | user@ip:port |
| 22:00.24 | Samot | That's it |
| 22:00.33 | sawgood | user, IP, port |
| 22:00.50 | Samot | When the other side sends and INVITE, it's going to AUTH by default. |
| 22:00.55 | Samot | That's how INVITES work |
| 22:01.17 | sawgood | unless you have invite=insecure,port |
| 22:01.24 | Samot | Or insecure=invite |
| 22:01.25 | [TK]D-Fender | host =ip = fucker IS there. Can't register. Calls refused from elsewhere. Has jack shit to do with PASWORD CHECKING |
| 22:01.34 | [TK]D-Fender | this is simple stuff |
| 22:01.39 | Samot | Right |
| 22:02.03 | [TK]D-Fender | Also says nothing about USERNAME checking |
| 22:02.05 | Samot | Knowing the location of the house doesn't mean the door is unlocked for you to walk in. |
| 22:02.12 | sawgood | permit= (allow) the username/password (to only come) from a specific IP |
| 22:02.16 | [TK]D-Fender | Which is mixed between the TYPE, and the action being taken |
| 22:02.35 | [TK]D-Fender | sawgood> permit= (allow) the username/password (to only come) from a specific IP <- no |
| 22:02.47 | [TK]D-Fender | Don't assume a username MATTERS yet |
| 22:03.18 | [TK]D-Fender | And depends on what the HOST setting is |
| 22:12.07 | *** part/#asterisk kharwell (kharwell@nat/digium/x-oxkshoibgbdycokt) |
| 22:15.27 | sawgood | if SIP traffic comes to Asterisk (and) if insecure=invite,port (is set) (so) no username/password is needed (what) does asterisk "use" to figure out which [context] to use (how does it now) |
| 22:15.42 | sawgood | know |
| 22:18.06 | [TK]D-Fender | Still wrong from the start |
| 22:18.23 | [TK]D-Fender | TYPE <- determines what gets checked. |
| 22:18.40 | [TK]D-Fender | Whether the username matters or not |
| 22:18.50 | [TK]D-Fender | Also depends on what ACTION is being taken |
| 22:19.08 | [TK]D-Fender | You don't seem to understand chan_sip at all |
| 22:20.11 | sawgood | I do: but I have a working process that has changed just slightly .... and other variables are involved which are hard to explain |
| 22:20.18 | [TK]D-Fender | "insecure" has nothing to do with an entry matching against the incoming call. It only affects whether a PASSWORD is checked for a CALL |
| 22:20.22 | [TK]D-Fender | You really don't |
| 22:20.32 | [TK]D-Fender | You don't understand the impact of any of these basics |
| 22:20.42 | [TK]D-Fender | you have no idea how matching actually works |
| 22:20.46 | sawgood | I worded my qustion wrong: I tried to word it in a way that your "one way" answer would fit |
| 22:21.07 | [TK]D-Fender | [TK]D-Fender> "insecure" has nothing to do with an entry matching against the incoming call. It only affects whether a PASSWORD is checked for a CALL |
| 22:21.12 | sawgood | I have a three lane road and your one-way answer is not always what I need (But it is very appreicated) |
| 22:21.28 | [TK]D-Fender | You keep asking about bits that aren't related |
| 22:21.45 | sawgood | I'm wording it wrong ... thats all ... give me a bit ... and I'll try in a differnt way |
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| 23:27.21 | hdon | hi all :) i'm looking at a channel identifier of the format C-[8HEXDIGITS] -- how can i identify which "Call ID" from "sip show channels" this is associated with? |
| 23:30.07 | Samot | You don't. |
| 23:30.24 | Samot | They aren't related outside of the fact they are two data points in the transaction. |
| 23:30.35 | Samot | Call ID is a SIP thing... |
| 23:30.36 | hdon | it looks like this "[C-%08x]" format string is part of logger.c and the value it's using comes from (at least sometimes; maybe all the time) from logmsg::callid |
| 23:30.43 | Samot | The C-XXXXXX is an Asterisk logging thing. |
| 23:30.43 | hdon | oh i see |
| 23:31.07 | Samot | Been like that for quite a long time. |
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| 23:34.57 | drmessano | Still on this, huh |