IRC log for #asterisk on 20170619

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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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08:14.25Kunsiif 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
13:33.14pawiecki"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-Fendernope.  You all merely drink coffee.  I was born with it.  Molded by it.
13:34.13[TK]D-Fenderhttps://uproxx.files.wordpress.com/2014/06/barista-bane-chai.jpg?quality=100&w=650&h=488
13:34.15pawieckithat sounds like an intro to some cool geeky comic
13:34.48[TK]D-Fender</justbanethings>
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14:00.41dnitHi 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 )
14:01.40dnitCurrently I am getting NoAnswer dialstatus for such calls
14:03.48Samotdnit: That's going to be based on the response from the carrier.
14:04.21Samotdnit: Depending on the carrier it could be a SIP code response or a ISDN code response.
14:05.14[TK]D-Fenderdnit, Nope
14:05.29Samotdnit: The OOS message you hear is Asterisk playing back that message based on the response of the carrier.
14:06.30SamotWell you can look at the HANGUPCAUSE
14:06.37SamotBut its not going to be 100%
14:06.59dnitSamot: 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
14:07.48SamotOK, if the carrier is the one playing back the message....
14:07.54SamotThat means it was answered.
14:07.54[TK]D-FenderForget about processing IB for this.
14:08.02SamotSo they could playback the message.
14:08.19SamotAggregate providers have to keep their stock "in service"
14:08.44SamotOr they risk getting "dead" stock pulled back by the OCN of the number.
14:10.52dnitSo is there nothing I can do to identify such calls, otherthan relying on HANGUPCAUSE
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14:14.11SamotIn Asterisk? Yes.
14:14.14SamotThat's it.
14:14.18SamotAnd that's not 100%
14:14.28Samothttps://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings
14:14.51SamotYou will note that multiple SIP codes are translated/mapped to 1 ISDN/Hangup cause
14:17.16dnitOh. Thanks for the info Samot and [TK]D-Fender
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14:20.04Samotdnit: 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.
14:20.11CrummyGummy@Samot Apologies, was AFK, I want to ring the one at a time.
14:20.42SamotCrummyGummy: Then you just need a Ring Group
14:21.14CrummyGummyHmmm
14:21.19CrummyGummyWill google
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14:22.33dnitSamot: The other side is answering before signalling 180, shouldn't that be considered differently ? ( I mean its the early media )
14:22.49Samot180 Ringing does not mean actual ringing
14:23.04SamotIt means the user (generally the phone number) is being attempted.
14:24.15dnitYes, but it atleast indicates the number is still active ?
14:24.31SamotActive in what way?
14:24.40SamotThat's YOUR provider sending back the 180
14:24.54SamotSaying "We're going to try and get this user"
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14:26.14dnitBy active I meant the provider doesn't know yet that the number is OOS.
14:26.27dnitand trying to reach it.
14:26.35SamotYou mean you're provider?
14:27.19dnitYes. two of my providers gave the similar OOS message inspite of SIP code.
14:27.30SamotOK.
14:27.31dnitThe carrier we have tied up with.
14:27.32SamotSo..
14:28.17SamotLet's assume it's PBX -> Carrier -> PSTN
14:28.19SamotOne hop
14:28.27SamotThere's probably more between you and the PSTN
14:28.38dnityes.
14:28.50SamotBut, when your call hits the Carrier, it's going to go "Is this my number? Do I own it?"
14:29.10SamotWhen it goes "No, I don't" it does a SPID lookup.
14:29.23SamotIt looks for who owns the number and where to route it to...
14:29.31SamotIf it finds a route, it sends it.
14:30.28SamotThe 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?"
14:31.18SamotSo you only know what your carrier tells you.
14:31.51SamotIn some cases they are telling you a direct response and in others they are relaying the response based off the other side.
14:33.30dnitSo 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 ) ?
14:34.07SamotCarriers only care if the DID has a route on the PSTN.
14:34.42SamotThe 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.
14:35.04SamotThat's the receiving carrier's issue.
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14:37.18SamotNow 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..
14:37.27SamotBut you could call them from your cell phone no issue.
14:37.40SamotThe carrier didn't have a path to that destination.
14:37.48SamotSwitched carriers and calls went through.
14:38.10SamotNot all carriers can connect calls to all destinations.
14:38.16SamotThey need a path to do it.
14:41.23dnitYes 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.
14:41.40SamotAsterisk is not a switch.
14:43.01SamotAnswered, No Answer, Busy, Congestion, Unavailable are pretty much the core responses Asterisk does.
14:44.43SamotWhat comes after the 183?
14:44.55dnitHmm
14:45.03dnitNothing only RTP
14:45.13SamotWhat SIP response?
14:45.22dnitthen after 45 seconds asterisk sends a cancel as I have 45 seconds timeout
14:45.37SamotWhat is after the 183 in regards to the SIP message.
14:45.41SamotIs it a 200 OK?
14:46.35dnitYes after asterisk sends a CANCEL 200 comes from carrier
14:46.52SamotSo the carrier sends a 183 followed by a 200?
14:47.03SamotAnd then Asterisk sends a CANCEL 45 seconds later?
14:47.08CrummyGummySamot: 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...
14:48.59dnitasterisk 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.
14:49.45SamotOK, that's one of two things.
14:49.58SamotAn upstream provider before the PSTN
14:50.15SamotOr a downstream provider after the PSTN
14:50.39SamotAsterisk is sending a CANCEL because there's no follow up response after the 183.
14:50.44SamotWhich there should be.
14:51.05SamotThen the carrier is OK'ing the CANCEL with that 200.
14:52.11SamotCan you call the number from your mobile?
14:52.22SamotThat one that has gotten two OOS messages from two different carriers?
14:55.18dnitNo I get the same message.
14:55.36SamotOK, the exact same message?
14:55.50dnitYes
14:55.56SamotThen it's the downstream.
14:56.13SamotThe (or a) carrier on the other side of the PSTN is sending that message.
14:56.23SamotAnd not clearing the call properly.
14:56.56SamotGenerally carriers just send back a code and let the other side deal with it.
14:57.16SamotIe the OOS message would be different per carrier.
14:57.51dnitFor few calls I got the same message for other calls I was able to call the number its a mix of both the situations .
14:59.21SamotWhat you are looking to do would need a switch/sip router in front of Asterisk.
14:59.39dnitYes I have the router.
14:59.53dnitusing kamailio
15:00.10SamotWell then you should be looking at these responses at that level.
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15:01.52dnitSorry 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 ?
15:02.08SamotClearly, this number is out of service.
15:02.25SamotHowever, a "real" carrier like ATT would not playback that message.
15:02.42SamotThey just would send back a code and let the other carrier, like Verizon, deal with it.
15:03.28SamotSmall providers, that get their numbers through aggregates or larger carriers needs to "answer" their calls.
15:04.04SamotTo show they are "using" those numbers and they are routing even if it's to a OOS or "Checking for trouble" message.
15:04.37SamotCarriers will snap back what they consider "dead" DIDs from their customers.
15:08.04dnitOh.
15:08.08SamotCarriers don't like "dead routes" for what they consider "assigned and routed" numbers. Skews their stats.
15:08.44SamotIt means those dead routes count as unanswered/failed calls to them.
15:10.57SamotPlus DID ownership is governed by a high body that the carriers have to answer to.
15:11.17Samots/high/higher/
15:12.23SamotYou should be looking at the SIP responses from the carriers at the Kamailio level and handle them accordingly..
15:12.47dnitOk got your point. Thanks for all the info.
15:12.56SamotThis case, however, shows the exception though.
15:13.25SamotWhere the carrier is actually "answering" the call and playing early media.
15:13.37SamotThat's probably going to slip by.
15:14.11SamotUnless you write some response timeout logic in Kamailio.
15:19.58dnitYes I got your whole point.
15:20.30dnitThanks for your time.
15:28.15SamotNp.
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18:46.48cervajs2@BenjaminKeithFord here?
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20:24.21TECFALLOutbound calls are dropping: chan_sip.c: Invalid contact uri  (missing sip: or sips:), attempting to use anyway
20:24.27TECFALLWhat is the issue?
20:24.55SamotShow a call this happens on
20:25.00Samotasterisk -rvvvvvvvvvv
20:25.05Samotsip set debug on
20:25.07Samot~pb
20:25.07infobotfrom 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.
20:26.09TECFALLSamot: ok, give me a minute.
20:28.29TECFALLcan i limit this by peer?
20:28.42TECFALLthere is a lot of packets flowing through
20:28.49Samotsip set debug peer <peername>
20:28.58Samotor sip set debug ip <ipaddress>
20:29.05SamotEither will get the same result.
20:30.35TECFALLand that can be an ipaddress of a sip phone right?
20:30.58SamotYes.
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20:34.14cervajs2can someone try replicate https://issues.asterisk.org/jira/browse/ASTERISK-27065 ? tnx
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