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| 14:31.27 | tiny4783 | Hello, |
| 14:31.27 | tiny4783 | <PROTECTED> |
| 14:31.27 | tiny4783 | some of them working fine. But another give this behaviour: |
| 14:31.27 | tiny4783 | incoming call's extensions changed to extension that has normal behaviour. |
| 14:32.13 | *** join/#asterisk scgm11_ (~scgm11@r186-49-10-244.dialup.adsl.anteldata.net.uy) |
| 14:32.14 | Samot | Wait, really Asterisk 1.6? |
| 14:32.22 | Samot | Or Asterisk 16? |
| 14:33.11 | *** join/#asterisk donuthole (~babbage_@80.111.186.87) |
| 14:33.45 | donuthole | How do I make sure that chan_sip is not enabled? |
| 14:34.43 | donuthole | I keep getting errors like this, despite the fact that I think I should be using pjsip instead: |
| 14:34.45 | donuthole | [Apr 11 15:34:06] NOTICE[8348]: chan_sip.c:28499 handle_request_register: Registration from '"6001 James" <sip:6001@telephony.spiral-arm.org>' failed for '192.168.15.207:5060' - Wrong password |
| 14:35.08 | file | you can noload it in modules.conf |
| 14:37.49 | Samot | Well if you're using PJSIP it's not on 5060 |
| 14:40.10 | tiny4783 | 1.6 |
| 14:41.00 | Samot | tiny4783: There is zero support for that |
| 14:42.03 | tiny4783 | Pc that hosted asterisk was terminated by blackout any option to clean cache maybe? |
| 14:42.54 | donuthole | Ah, I had temporarily unconfigured the UDP transport. I had reversed that and reloaded, but that doesn't work in pjsip. Fixed now. |
| 14:43.29 | file | PJSIP isn't really written to have dynamic transport configuration like that, at least safely |
| 14:43.56 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 14:46.15 | [TK]D-Fender | <tiny4783> 1.6 <- 1.6 is ancient and PJSIP didn't exist... |
| 14:46.46 | [TK]D-Fender | tiny4783, Strong suspicion you're mixing it up with 16 |
| 14:46.59 | tiny4783 | nope definitly have 1.6 |
| 14:47.07 | seanbright | he hasn't mentioned pjsip |
| 14:47.08 | tiny4783 | we are in migration to 16 |
| 14:47.47 | Samot | [TK]D-Fender: You're crossing two ongoing conversations with each other. |
| 14:47.56 | [TK]D-Fender | yup |
| 14:48.02 | [TK]D-Fender | my bad... |
| 14:48.11 | [TK]D-Fender | haven't finished that first coffee today |
| 15:00.55 | donuthole | I'm getting "403 Forbidden" from my ITSP in response to INVITE requests. See paste of debug info at https://pastebin.com/PXJNSQm7 |
| 15:01.21 | donuthole | [TK]D-Fender, you told me the other day that this is because I have an un-routable address in the outgoing Contact header. How do I change that? |
| 15:01.28 | donuthole | (is that the right fix?) |
| 15:01.44 | [TK]D-Fender | Contact: <sip:asterisk@192.168.122.105:5060> |
| 15:01.48 | [TK]D-Fender | indeed still bad |
| 15:02.11 | donuthole | How do I put a more useful address in there? |
| 15:02.22 | [TK]D-Fender | you have to tell * your WAN IP, define your localnet's, prevent directmedia, etc |
| 15:02.46 | [TK]D-Fender | externaddr,localnet,directmedia <- these under [general] |
| 15:02.57 | donuthole | Ack. Let's fix the control layer first. My WAN IP is dynamic (though slowly changing) |
| 15:03.20 | donuthole | I have a local TURN & STUN server which should be working (though it would have no clients other than Asterisk) |
| 15:03.56 | donuthole | [general] in pjsip.conf? |
| 15:04.56 | [TK]D-Fender | Sorry, still in that previous mixup. Those were chan_sip settings |
| 15:05.02 | [TK]D-Fender | go read the sample config |
| 15:05.13 | [TK]D-Fender | The parameters are very similar |
| 15:05.21 | [TK]D-Fender | and are defined in your transport |
| 15:05.30 | donuthole | I'm using chan_pjsip. The documentation seemed to imply that it was a better alternative. |
| 15:05.48 | donuthole | So for pjsip I set this in [transport-udp] and [transport=tcp]? |
| 15:06.59 | [TK]D-Fender | in your transport |
| 15:07.03 | [TK]D-Fender | whatever you called it |
| 15:07.33 | seanbright | donuthole: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT |
| 15:09.42 | Samot | 11:02:46 AM <[TK]D-Fender> externaddr,localnet,directmedia <- these under [general] <-- This doesn't exist in PJSIP |
| 15:09.58 | Samot | There are no global general options for PJSIP endpoints. |
| 15:10.00 | Samot | Or transports. |
| 15:10.30 | [TK]D-Fender | Samot, already corrected |
| 15:10.51 | [TK]D-Fender | Samot, carryover from my previous mixup |
| 15:11.19 | Samot | donuthole: Do you use SIP Registration/Auth with your provider? |
| 15:11.53 | donuthole | Yes, the registration seems to be workign already. |
| 15:12.03 | donuthole | Likewise auth |
| 15:12.06 | Samot | OK so the 403 only happens on the INVITE? |
| 15:12.10 | donuthole | yes |
| 15:13.56 | Samot | Do they want you to specify a specific from user or domain? |
| 15:14.43 | donuthole | Yes, but I have that correct now. |
| 15:15.12 | Samot | Show a full failed call |
| 15:15.23 | Samot | From the first INVITE to the 403 response. |
| 15:15.34 | Samot | that means there should be a 401 in there as well. |
| 15:16.27 | donuthole | OK, so the Contact address now lists the correct address (for right now, the current DHCP least). But the port number is 5060, and there is a SIP phone on the internal network here which the same ITSP thinks is listening on that address:port (80.111.186.87:5060>) |
| 15:17.02 | donuthole | I'll just change the DNAT rule to send port 5060 traffic to the Asterisk server for now. One moment... |
| 15:22.01 | donuthole | Here is a full call (which is originated at a local SIP phone) https://pastebin.com/FdzpvaTH |
| 15:22.50 | donuthole | And yes, it contains two 401s. One from Asterisk to the SIP handset, and the other from the ITSP to Asterisk. |
| 15:24.28 | Samot | That means you failed auth. |
| 15:25.36 | Samot | Initial attempt, 401 is accepted. They didn't like the details, a second 401 was sent. |
| 15:26.00 | Samot | They didn't like what you sent back to that second 401 and they sent back a 403 and denied you. |
| 15:26.40 | Samot | s/accepted/expected/ |
| 15:28.28 | donuthole | Ah, I'm sure it had been working. |
| 15:28.49 | donuthole | Let me re-check the realm data |
| 15:29.01 | Samot | You'll need to ask the provider. |
| 15:30.16 | *** join/#asterisk scgm11_ (~scgm11@r186-49-10-244.dialup.adsl.anteldata.net.uy) |
| 15:30.31 | [TK]D-Fender | From: "6001 James" <sip:6001@192.168.122.105>;tag=ed81e370-0e91-4720-890b-7fa347b8add5 |
| 15:30.54 | [TK]D-Fender | I do wonder if they don't appreciate you presenting the "from" as forwarded from the calling channel |
| 15:40.02 | *** join/#asterisk qakhan (~qakhan@50-204-254-11-static.hfc.comcastbusiness.net) |
| 15:42.40 | qakhan | hi all, there are 3 queues. q1,q21 and q3 and 3 agents. all 3 agents are members in 3 qeueus. now call comes in for q2 then q3 then q1. |
| 15:42.56 | qakhan | when Q call will ring first |
| 15:43.25 | donuthole | Samot, you are right, the REGISTER is actually not working. |
| 15:43.28 | qakhan | which* |
| 15:44.03 | qakhan | all Qs ring strategy is "Ringall" |
| 15:48.13 | *** join/#asterisk scgm11_ (~scgm11@r186-49-10-244.dialup.adsl.anteldata.net.uy) |
| 15:49.43 | donuthole | OK, "pjsip show regstrations" now shows that the registration works, but the INVITE still 403s. |
| 15:49.43 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 15:54.14 | *** join/#asterisk juser123 (~juser123@d28-23-130-66.dim.wideopenwest.com) |
| 15:54.20 | [TK]D-Fender | qakhan, Whichever got the call first |
| 16:09.44 | *** join/#asterisk nixfreak (324d36de@gateway/web/freenode/ip.50.77.54.222) |
| 16:12.36 | nixfreak | Hello , I used to run an Asterisk box about a decade ago though , I trying to find information on how to route cell phone calls through an asterisk box for spam purposes and only give out a DID that I can bind to my cell phone |
| 16:13.39 | nixfreak | So route cell phone calls through through the box and have it automatically get dumpe |
| 16:14.01 | [TK]D-Fender | Calls to your phone go to your phone.... |
| 16:14.35 | donuthole | FFS. The ITSP seems to be using different auth realms in their WWW-Authenticate challenges for REGISTER and INVITE. Adding in a comma-separated list of auth items so that one matches each realm still results in a 403 for the INVITE part of the session. I think maybe I should just change ITSP. |
| 16:14.48 | [TK]D-Fender | if you're looking to get another # and give that out and let * make the decision to send a caller on to your cell (or not), then that's something else which * can do |
| 16:20.22 | nixfreak | If someone calls my number , I would like * to handle it yes, but can't I get a DID for others to have that route to my number ? |
| 16:20.44 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 16:22.25 | nixfreak | [TK]D-Fender where can I get information on that than ? |
| 16:30.18 | [TK]D-Fender | if they call your cell # then it goes to your cell |
| 16:30.45 | [TK]D-Fender | If they call another number you get from a provider that your server connects to then it goes to your server |
| 16:31.01 | [TK]D-Fender | And from there you can do whatever you want from that point. |
| 16:34.44 | nixfreak | but can't I forward my cell to the * box ? |
| 16:38.27 | *** join/#asterisk troyt (zncsrv@2601:681:4100:8981:44dd:acff:fe85:9c8e) |
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| 16:42.57 | [TK]D-Fender | You'll need to forward it to another number which you'll almost certainly have to pay for. |
| 16:43.14 | [TK]D-Fender | Then it'll also have to be able to call back out |
| 16:43.27 | [TK]D-Fender | But your phone is FORWARDED. How do you expect that call to make it through? |
| 16:44.29 | nixfreak | No I get it its forwarded |
| 16:45.00 | [TK]D-Fender | I see no way to try to force * in betweent hese two transparently |
| 16:46.40 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 16:50.35 | nixfreak | My goal was / is to route all calls to Asterisk and dump the calls unless you call into the server using a code which than routes back to the phone number |
| 16:50.51 | nixfreak | sort of like a switch |
| 16:51.02 | nixfreak | I mean router |
| 16:51.59 | [TK]D-Fender | Except I'm not seeing a path back |
| 16:52.25 | [TK]D-Fender | not using the number on the cell as the initial point of contact |
| 16:52.45 | [TK]D-Fender | Using a separate number and distributing that instead? Sure. |
| 16:53.49 | nixfreak | hmm |
| 16:56.45 | nixfreak | ok I with that first than and see If I can figure out another way with the single number. Would the 4th edition book have the details for that , or do I need to read something else. The last time I ran a asterisk box was for phone conferencing |
| 16:57.54 | [TK]D-Fender | "see If I can figure out another way with the single number." <- not happening |
| 16:58.17 | [TK]D-Fender | There are no details needed for this |
| 16:58.31 | [TK]D-Fender | nor any guide. There is no need for this. |
| 16:59.06 | [TK]D-Fender | Call arrives. You look at it and decide if you want to let it through. If so just dial out. |
| 16:59.13 | [TK]D-Fender | That's it. |
| 16:59.49 | [TK]D-Fender | Nobody makes a guide for could be 3 lines of dialplan. |
| 17:01.56 | nixfreak | ok |
| 17:03.12 | donuthole | quit |
| 17:03.15 | donuthole | doh |
| 17:03.19 | [TK]D-Fender | quite |
| 17:04.01 | [TK]D-Fender | #quiet(er) |
| 17:11.29 | *** part/#asterisk nixfreak (324d36de@gateway/web/freenode/ip.50.77.54.222) |
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| 18:55.35 | *** join/#asterisk anexit (~anexit@push.anexit.net) |
| 18:55.43 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:55.45 | anexit | Can anyone explain this; a=X-nat:0 |
| 18:55.58 | anexit | I know it's nat but does that mean the sdp is not using it? |
| 19:01.13 | *** join/#asterisk rpifan (~rpifan@dslb-178-000-177-094.178.000.pools.vodafone-ip.de) |
| 19:11.02 | seanbright | https://www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml |
| 19:11.08 | seanbright | it's non-standard, so who nows |
| 19:11.10 | seanbright | knows* |
| 19:11.46 | anexit | huh |
| 19:12.11 | seanbright | is it a "UniFi VoIP Phone?" |
| 19:12.20 | anexit | Switchvox |
| 19:12.40 | seanbright | pjsip adds it apparently |
| 19:13.59 | seanbright | 0 means "unknown" |
| 19:14.10 | anexit | Where did you see that? |
| 19:14.24 | sawgood | Hi: is this correct: If I want a PJSIP trunk (from) Asterisk 16 over to another Asterisk 16 box: will I need to have (all 6) PJSIP objects in the (2) pjsip.conf files (one for each Asterisk box)? |
| 19:14.40 | sawgood | for a registration type PJSIP trunk (sorry) forgot to mention that |
| 19:14.44 | seanbright | https://trac.pjsip.org/repos/browser/pjproject/trunk/pjnath/src/pjnath/nat_detect.c#L33 |
| 19:14.47 | seanbright | anexit: ^^ |
| 19:14.57 | seanbright | the number is a 0-based offset into that array |
| 19:17.22 | seanbright | anexit: nothing should be paying attention to that attribute, so its value is pretty much meaningless |
| 19:18.50 | Samot | sawgood: Do you control both boxes? |
| 19:19.19 | anexit | seanbright: thanks! |
| 19:33.14 | *** join/#asterisk jkroon (~jkroon@165.16.203.106) |
| 19:52.31 | sawgood | Samot: yes I do |
| 19:52.45 | Samot | Then why is registration needed? |
| 19:52.49 | sawgood | Asterisk 16 on both sides: they are NOT on the same LAN |
| 19:53.02 | Samot | OK are they over NAT? |
| 19:53.09 | sawgood | one is a LAN behind NAT: one is WAN-only with a public static IP |
| 19:53.26 | Samot | Which is which? |
| 19:53.40 | sawgood | both are Asterisk 16-only PJSIP boxes |
| 19:53.41 | Samot | Which of those is the one accepting the inbound registration? |
| 19:53.56 | Samot | The WAN one? |
| 19:53.57 | sawgood | the one taking IN the registration is the WAN-side box |
| 19:54.25 | sawgood | maybe it would be easier if I did a IP peer trunk 1st: I don't mind taking that route |
| 19:54.32 | Samot | So then you need to set up an endpoint/aor/auth/identity (if needed) on the WAN box |
| 19:54.40 | sawgood | I meant I could do an IP peer instead |
| 19:54.48 | Samot | And do the same but use outbound_auth and registration on the other box. |
| 19:55.05 | sawgood | nice: thank you: in progress ... I'll be back |
| 19:55.07 | Samot | Only if you control the router/firewall and can setup the proper NAT rules. |
| 19:55.22 | sawgood | I do |
| 19:56.07 | sawgood | but: not to change the subject: I have this same LAN-only Asterisk 16 box: with a FPBX box PJSIP on the same LAN: and I couln't get a registration type trunk to work: so since they are on the SAME LAN: I'll make them an IP peer trunk ... |
| 20:01.17 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
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| 20:56.02 | life_of_e | How much work is Digium/Sangoma putting into fax support in Asterisk? Is that a back burner feature or is that something that has a relatively constant view? |
| 20:58.25 | *** join/#asterisk stefanauss (~stefanaus@160.97.54.39) |
| 20:58.47 | file | it's not something actively developed or expanded, if issues come up then they will go into queue |
| 21:03.54 | life_of_e | Ok, I asked because HylaFax just released an update supporting SSL Fax which requires specific signaling initially. I suspect that Asterisk would fall out of the way after detecting the tone and passing on to the modem and HylaFax. |
| 21:04.40 | file | the Hylafax work is something done externally, there's nothing in Asterisk itself in regards to it specifically - only the general stuff |
| 21:04.48 | file | so - dunno |
| 21:06.16 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 21:09.30 | life_of_e | Yeah, I'm thinking it'll just be a pass-off after the tone detection and the handshake would happen after HylaFax picks up the modem. |
| 21:10.55 | life_of_e | It's an interesting concept, using the audio channel as the signaling path to transmit a URL for an OOB data transmission with fallback to inband data transmission. |
| 21:11.03 | life_of_e | https://hylafax.sourceforge.io/sslfax.php |
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| 22:51.55 | bipolar | Has anyone here ever had RDNIS missing only from certain calls? In my case, I set my Verizon wireless cell phone to forward missed calls to my Asterisk DID. If I call my cell phone from my Google Voice account, I get the RDNIS when it gets forwarded. But if I call from another Verizon wireless phone, I don't get RDNIS at all. I've checked this at the PRI level. I'm hoping someone has some experience with this and can point me in the right direction. |
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| 23:23.15 | Samot | bipolar: Google Voice is nothing more than a Call Forwarding service if you don't use their apps. |
| 23:23.28 | Samot | RDNIS is presented for redirected calls, which is what GV is doing. |
| 23:24.09 | Samot | Have you tested with a non-Verizon service? |
| 23:24.25 | Samot | Do you get RDNIS when you use ATT or another carrier to call your phone? |
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| 23:28.30 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.26.0 (2019/4/4) 16.3.0 (2019/4/4), Security Only: 15.7.2 (2019/2/28); DAHDI: 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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