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00:18.33 | *** 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 |
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08:14.25 | Kunsi | if my asterisk server is behind nat, but the remote sip server is not, do i have to set 'nat=yes' in sip.conf (on local end)? |
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13:33.08 | *** 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 |
13:33.14 | pawiecki | "It takes me 1 minute everyday to make a cup of coffee... And now since i'm using asterisk a lot, why don't i spend months of my life to make AI to do it for me" - sounds worthwile |
13:33.43 | [TK]D-Fender | nope. You all merely drink coffee. I was born with it. Molded by it. |
13:34.13 | [TK]D-Fender | https://uproxx.files.wordpress.com/2014/06/barista-bane-chai.jpg?quality=100&w=650&h=488 |
13:34.15 | pawiecki | that sounds like an intro to some cool geeky comic |
13:34.48 | [TK]D-Fender | </justbanethings> |
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14:00.41 | dnit | Hi guys, Is there a way in asterisk where we can detect the call as number out of service as dialstatus ( RTP saying that the number is out of service in early media ) |
14:01.40 | dnit | Currently I am getting NoAnswer dialstatus for such calls |
14:03.48 | Samot | dnit: That's going to be based on the response from the carrier. |
14:04.21 | Samot | dnit: Depending on the carrier it could be a SIP code response or a ISDN code response. |
14:05.14 | [TK]D-Fender | dnit, Nope |
14:05.29 | Samot | dnit: The OOS message you hear is Asterisk playing back that message based on the response of the carrier. |
14:06.30 | Samot | Well you can look at the HANGUPCAUSE |
14:06.37 | Samot | But its not going to be 100% |
14:06.59 | dnit | Samot: Inspecting the pcaps I fould there was no SIP response code (such as 480 481) there is only 183 session in progress response prior to message from carrier saying number is out of service |
14:07.48 | Samot | OK, if the carrier is the one playing back the message.... |
14:07.54 | Samot | That means it was answered. |
14:07.54 | [TK]D-Fender | Forget about processing IB for this. |
14:08.02 | Samot | So they could playback the message. |
14:08.19 | Samot | Aggregate providers have to keep their stock "in service" |
14:08.44 | Samot | Or they risk getting "dead" stock pulled back by the OCN of the number. |
14:10.52 | dnit | So is there nothing I can do to identify such calls, otherthan relying on HANGUPCAUSE |
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14:14.11 | Samot | In Asterisk? Yes. |
14:14.14 | Samot | That's it. |
14:14.18 | Samot | And that's not 100% |
14:14.28 | Samot | https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings |
14:14.51 | Samot | You will note that multiple SIP codes are translated/mapped to 1 ISDN/Hangup cause |
14:17.16 | dnit | Oh. Thanks for the info Samot and [TK]D-Fender |
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14:20.04 | Samot | dnit: But again, if the other side is answering and playing back a message, the call might not show up as anything but a completed call to Asterisk. |
14:20.11 | CrummyGummy | @Samot Apologies, was AFK, I want to ring the one at a time. |
14:20.42 | Samot | CrummyGummy: Then you just need a Ring Group |
14:21.14 | CrummyGummy | Hmmm |
14:21.19 | CrummyGummy | Will google |
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14:22.33 | dnit | Samot: The other side is answering before signalling 180, shouldn't that be considered differently ? ( I mean its the early media ) |
14:22.49 | Samot | 180 Ringing does not mean actual ringing |
14:23.04 | Samot | It means the user (generally the phone number) is being attempted. |
14:24.15 | dnit | Yes, but it atleast indicates the number is still active ? |
14:24.31 | Samot | Active in what way? |
14:24.40 | Samot | That's YOUR provider sending back the 180 |
14:24.54 | Samot | Saying "We're going to try and get this user" |
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14:26.14 | dnit | By active I meant the provider doesn't know yet that the number is OOS. |
14:26.27 | dnit | and trying to reach it. |
14:26.35 | Samot | You mean you're provider? |
14:27.19 | dnit | Yes. two of my providers gave the similar OOS message inspite of SIP code. |
14:27.30 | Samot | OK. |
14:27.31 | dnit | The carrier we have tied up with. |
14:27.32 | Samot | So.. |
14:28.17 | Samot | Let's assume it's PBX -> Carrier -> PSTN |
14:28.19 | Samot | One hop |
14:28.27 | Samot | There's probably more between you and the PSTN |
14:28.38 | dnit | yes. |
14:28.50 | Samot | But, when your call hits the Carrier, it's going to go "Is this my number? Do I own it?" |
14:29.10 | Samot | When it goes "No, I don't" it does a SPID lookup. |
14:29.23 | Samot | It looks for who owns the number and where to route it to... |
14:29.31 | Samot | If it finds a route, it sends it. |
14:30.28 | Samot | The owner gets it and does the same thing pretty much "Oh is this my number being sent to me? It is? Where do I send it?" |
14:31.18 | Samot | So you only know what your carrier tells you. |
14:31.51 | Samot | In some cases they are telling you a direct response and in others they are relaying the response based off the other side. |
14:33.30 | dnit | So you are saying my carrier itself doesnt know that the number is OOS based on the OOS message ( due to lack of particular SIP response ) ? |
14:34.07 | Samot | Carriers only care if the DID has a route on the PSTN. |
14:34.42 | Samot | The carrier sending the calls doesn't care if the receiving carrier doesn't have an internal route or if that number is routed or in use on their network. |
14:35.04 | Samot | That's the receiving carrier's issue. |
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14:37.18 | Samot | Now I just had an issue in my LCR routing that was sending calls to a certain destination out a carrier and the calls got OOS messages.. |
14:37.27 | Samot | But you could call them from your cell phone no issue. |
14:37.40 | Samot | The carrier didn't have a path to that destination. |
14:37.48 | Samot | Switched carriers and calls went through. |
14:38.10 | Samot | Not all carriers can connect calls to all destinations. |
14:38.16 | Samot | They need a path to do it. |
14:41.23 | dnit | Yes I understand that. But my point is how to catch this OOS in asterisk and provide an appropriate status to that particular call ( based on carriers OOS message ) as it seems to be a pretty common scenario. |
14:41.40 | Samot | Asterisk is not a switch. |
14:43.01 | Samot | Answered, No Answer, Busy, Congestion, Unavailable are pretty much the core responses Asterisk does. |
14:44.43 | Samot | What comes after the 183? |
14:44.55 | dnit | Hmm |
14:45.03 | dnit | Nothing only RTP |
14:45.13 | Samot | What SIP response? |
14:45.22 | dnit | then after 45 seconds asterisk sends a cancel as I have 45 seconds timeout |
14:45.37 | Samot | What is after the 183 in regards to the SIP message. |
14:45.41 | Samot | Is it a 200 OK? |
14:46.35 | dnit | Yes after asterisk sends a CANCEL 200 comes from carrier |
14:46.52 | Samot | So the carrier sends a 183 followed by a 200? |
14:47.03 | Samot | And then Asterisk sends a CANCEL 45 seconds later? |
14:47.08 | CrummyGummy | Samot: I got it working like you said thanks, I'd tried that before but it didn't work. I think I had something else wrong at the time... |
14:48.59 | dnit | asterisk sends INVITE , Carrier sends 100 Trying , then carrier sends 183 trying, after 45 seconds asterisk sends CANCEL , carrier sends 200 OK , carrier sends 487 termination. |
14:49.45 | Samot | OK, that's one of two things. |
14:49.58 | Samot | An upstream provider before the PSTN |
14:50.15 | Samot | Or a downstream provider after the PSTN |
14:50.39 | Samot | Asterisk is sending a CANCEL because there's no follow up response after the 183. |
14:50.44 | Samot | Which there should be. |
14:51.05 | Samot | Then the carrier is OK'ing the CANCEL with that 200. |
14:52.11 | Samot | Can you call the number from your mobile? |
14:52.22 | Samot | That one that has gotten two OOS messages from two different carriers? |
14:55.18 | dnit | No I get the same message. |
14:55.36 | Samot | OK, the exact same message? |
14:55.50 | dnit | Yes |
14:55.56 | Samot | Then it's the downstream. |
14:56.13 | Samot | The (or a) carrier on the other side of the PSTN is sending that message. |
14:56.23 | Samot | And not clearing the call properly. |
14:56.56 | Samot | Generally carriers just send back a code and let the other side deal with it. |
14:57.16 | Samot | Ie the OOS message would be different per carrier. |
14:57.51 | dnit | For few calls I got the same message for other calls I was able to call the number its a mix of both the situations . |
14:59.21 | Samot | What you are looking to do would need a switch/sip router in front of Asterisk. |
14:59.39 | dnit | Yes I have the router. |
14:59.53 | dnit | using kamailio |
15:00.10 | Samot | Well then you should be looking at these responses at that level. |
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15:01.52 | dnit | Sorry did not understand what you mean by looking at the response at that level. Should I try to attemp the number via another carrier at router level if such situations happen ? |
15:02.08 | Samot | Clearly, this number is out of service. |
15:02.25 | Samot | However, a "real" carrier like ATT would not playback that message. |
15:02.42 | Samot | They just would send back a code and let the other carrier, like Verizon, deal with it. |
15:03.28 | Samot | Small providers, that get their numbers through aggregates or larger carriers needs to "answer" their calls. |
15:04.04 | Samot | To show they are "using" those numbers and they are routing even if it's to a OOS or "Checking for trouble" message. |
15:04.37 | Samot | Carriers will snap back what they consider "dead" DIDs from their customers. |
15:08.04 | dnit | Oh. |
15:08.08 | Samot | Carriers don't like "dead routes" for what they consider "assigned and routed" numbers. Skews their stats. |
15:08.44 | Samot | It means those dead routes count as unanswered/failed calls to them. |
15:10.57 | Samot | Plus DID ownership is governed by a high body that the carriers have to answer to. |
15:11.17 | Samot | s/high/higher/ |
15:12.23 | Samot | You should be looking at the SIP responses from the carriers at the Kamailio level and handle them accordingly.. |
15:12.47 | dnit | Ok got your point. Thanks for all the info. |
15:12.56 | Samot | This case, however, shows the exception though. |
15:13.25 | Samot | Where the carrier is actually "answering" the call and playing early media. |
15:13.37 | Samot | That's probably going to slip by. |
15:14.11 | Samot | Unless you write some response timeout logic in Kamailio. |
15:19.58 | dnit | Yes I got your whole point. |
15:20.30 | dnit | Thanks for your time. |
15:28.15 | Samot | Np. |
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18:46.48 | cervajs2 | @BenjaminKeithFord here? |
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20:24.21 | TECFALL | Outbound calls are dropping: chan_sip.c: Invalid contact uri (missing sip: or sips:), attempting to use anyway |
20:24.27 | TECFALL | What is the issue? |
20:24.55 | Samot | Show a call this happens on |
20:25.00 | Samot | asterisk -rvvvvvvvvvv |
20:25.05 | Samot | sip set debug on |
20:25.07 | Samot | ~pb |
20:25.07 | infobot | from memory, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
20:26.09 | TECFALL | Samot: ok, give me a minute. |
20:28.29 | TECFALL | can i limit this by peer? |
20:28.42 | TECFALL | there is a lot of packets flowing through |
20:28.49 | Samot | sip set debug peer <peername> |
20:28.58 | Samot | or sip set debug ip <ipaddress> |
20:29.05 | Samot | Either will get the same result. |
20:30.35 | TECFALL | and that can be an ipaddress of a sip phone right? |
20:30.58 | Samot | Yes. |
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20:34.14 | cervajs2 | can someone try replicate https://issues.asterisk.org/jira/browse/ASTERISK-27065 ? tnx |
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23:52.57 | *** join/#asterisk lankanmon_ (~LKNnet@2607:fea8:d1f:ffcb:f9bc:b6f9:564b:9fd4) |