IRC log for #asterisk on 20210204

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16:18.38MLCSeems like the QUEUEHOLDTIME variable is in seconds. Can anyone confirm or deny?
16:24.39seanbrightconfirmed
16:24.43SamotYes
16:24.45MLCthanks
16:24.52seanbrighti will not be taking follow ups
16:25.08MLCdarn, I was going to ask about the meaning of life as a followup
16:25.09SamotThanks sean spencer
16:26.57SamotOr Spicer...yeah shows how fast I forgot about him
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16:38.51vasilakisfilwhat do you think about SIP ? will it be relevant in 10 years? from what I hear it's a bit difficult to scale sip in the era of cloud.. Personally I doubt that any other protocol will manage to replace all SIP features/rfcs
16:40.20Samot11:39:17 AM <Samot> What?
16:40.20Samot11:39:23 AM <Samot> How is it difficult to scale?
16:40.28SamotFrom my reply in #kamailio
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16:45.47igcewielingI think SIP beat out IAX, IAX2, MGCP, SCCP (arguably) , and H323.   I'm not worried about it going away.
16:46.18igcewielingPersonally, wish MGCP or IAX2 won the protocol wars, but SIP won.
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16:53.55igcewieling*sigh*  Apparently, our Hosted platform returns a 486 when a destination is busy AND when a DID is not loaded.   So, no way to tell if the call failed because someone didn't do their job and load the number or if the extension was just busy without voicemail setup.
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18:25.37vanczHi folks. I know nothing about telephony internals, nor about PBX's but I had an idea and I'm wondering how feasible it is or where I could go to learn more; I know for voice calls you can get multiplexed to internal phone numbers, could you do the same with SMS?
18:26.18vanczOh hm. I just realized when you do the voice multiplexing you have to dial the extension after the pbx picks up, so its just doing it at the software level using the number tones
18:26.21vancz:I
18:27.26vanczHm, unrelatedly, how can a pbx receive multiple simultaneous calls? or is that somehow solved by using IP these days
18:29.01SamotIt's been solved for decades and before IP
18:29.14vanczOk, I wasnt sure about that
18:29.50vanczI was wondering if it would be possible to make a temporary sms service simlar to the email ones - getting numbers seems difficult/expensive? and I imagine phone number space is relatively limited. So I was wondering if you could have a handful of numbers and do some multiplexing with the suffix as a sort of preshared key
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18:41.29SamotSure if you want to build it
18:41.41SamotBut numbers arent the issue
18:45.01vanczSamot: come on you cant stop there, do tell :p
18:45.32SamotWhat do you mean?
18:45.40vanczwhat *is* the issue
18:45.52SamotThere isn't one
18:46.08SamotThere is not issue that requires your solutuon
18:46.12SamotSolution
18:46.26vanczok
18:47.03SamotThere isn't a shortage of DIDs
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19:15.20DerKohi, im trying to create a call using /var/spool/asterisk/outgoing/ that will stay active until the called party hangs up
19:15.54seanbrightsounds good
19:16.01seanbrightlet us know if we can help
19:17.13seanbrightcalls will stay up until one party hangs up by default
19:17.25seanbrightso if that is not happening you are doing something to cause it
19:17.45DerKothis is what my call file looks like
19:17.46DerKohttps://pastebin.com/vXmPV5n0
19:18.09SamotDoes the call work?
19:18.10DerKothe extentions.conf is basically just a dial
19:18.12DerKoyes
19:18.21SamotSo what is the problem?
19:18.22DerKobut it cancels 10 seconds after the 200 ok
19:18.36DerKoi would like for it to stay active a bit longer
19:18.41SamotOK, that could be a number of things
19:18.46SamotNot related to the call file.
19:19.09DerKowhan i make a manual call
19:19.11DerKoit does not happen
19:19.20DerKoand the cancel is initiated by asterisk
19:19.37DerKoso i figured is the way this outbound module works
19:19.42DerKoand something im doing wrong
19:20.26SamotWell we need to see a full debug of the outgoing call
19:20.33Samotasterisk -rvvvvvvvvvvvv
19:20.40Samotsip set debug on
19:20.45Samotthen trigger the callfile
19:21.42DerKodoing it
19:24.33DerKohttps://pastebin.com/bjKsphgt
19:29.39igcewieling[Feb  4 19:23:05] WARNING[2462][C-000021a5]: pbx.c:4510 __ast_pbx_run: Channel 'SIP/10.116.248.205:5160-000026b3' sent to invalid extension but no invalid handler: context,exten,priority=other,3000,1
19:32.25seanbrightyup
19:32.36seanbrightthe WARNING was a dead giveaway
19:45.00vanczwww.ozekisms.com/p_2547-ozeki-1.4-sms-number-formats.html#1.4.5 "Long code means that the telephone number is longer than the standard phone number length in the network. e.g.: +36201234567111111In this case, a freely defined postfix is appended to the phone number itself. This feature is available in some networks only. It is very useful because the postfix part of the telephone number can by used to carry a unique message id, that can be used by
19:45.00vanczapplications. [..]  Long code numbers are only available on IP SMS connections (such as SMPP, UCP, CIMD2). The GSM modem technology does not permit this. If you use a GSM modem, you can only receive messages on the appropriate MSISDN number."
19:46.02SamotWhat?
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20:12.07igcewielingvancz: that is not supported in the USA or Canada
20:13.00igcewielingSomething which sounds identical has been available in at least Germany on PRIs for at least a decade.
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20:32.26Kobazsooooooo
20:33.26Kobazin pjsip, is there a way to try and match an endpoint if an identify matches but doesn't pass?
20:34.21KobazIe: I have an endpoint configured in pjsip-wizard called say.. '205',  and i just so happen to have an identify that matches the ip address that a REGISTER is coming from... but does not match the auth associated
20:35.31Kobazso for example... https://dpaste.com/CMSM89722
20:35.57Kobazso.. yes is-bob-main-identify does match the ip address in question.. but there is a *better* match, that accepts registrations
20:39.30Kobazwhich is this one: https://dpaste.com/8MLTM6LVV
20:45.41Kobazmy issue is... from the log:  this REGISTER should be matching the type=wizard endpoint.  not those other identifies
20:46.04SamotWhere is that ebdpoint associated with the identity?
20:46.33KobazIt's not
20:46.33SamotAnd is that endpoints being told to check that vs user/ip
20:47.08SamotThere is an author order
20:47.17SamotAuth
20:47.22Kobazah
20:47.37Kobazis-bob-main is another endpoint wizard
20:47.37SamotUser/IP is default
20:47.43Kobazwith remote hosts
20:47.56Kobazokay... i have my chan_sip set up for auth_name_match
20:48.04Kobazso it prioritizes endpoint names that match, and THEN ips
20:48.11Kobazbut pjsip is.... doing... not sure what
20:48.35igcewielingit does what endpoint_idenfity_order is set to, IIRC
20:48.43SamotCorrect
20:48.53SamotUser/IP is default
20:48.59Kobazk
20:48.59Kobazcool
20:49.03SamotThat is checked first
20:49.18Kobazendpoint_idenfity_order is... global?
20:49.24Kobazlet me look that up
20:49.25igcewielingKobaz: I ran into something similar yesterday.
20:49.39SamotNo
20:49.45SamotIt is per endpoint
20:50.36igcewielingI set up an [anonymous] endpoint to catch bots.  But one of my peers, one with a static IP send some calls to me from anonymous and it matched the anonymous endpoint instead of the correct one.   Since I don't care that much about messing with bots, I remove the anonymous endpoint.
20:51.16Kobazright
20:51.24Kobazsince names, can cause issues like that
20:51.50Kobazi had that exact issue because i had say a peer in chan_sip called '101'
20:51.55igcewielingKobaz: it was sending "anonymous" on calls without callerid blocked.   evil.
20:52.23KobazAnd then, a pbx we were integrating with, was sending '101 as a callerid for transferred calls, like a blind transfer from the real 101
20:52.28Kobazand then it was failing auth
20:52.43igcewielingKobaz: the main reason I switched to pjsip to get away from chan_sip's incredibly confusing matching/auth.   pjsip might be more complicated, but it makes sense, most of the time.
20:53.16Kobazsince the call from '101 was coming over a trunk from a pbx, it should always auth
20:53.36Kobazbut my issue i ran into with the match name first, was... blind transfers set the From: and then causes it to fail auth
20:53.36SamotNope
20:53.49SamotThe first peer to match wins
20:54.04SamotIf 101 exist s before the trunk, it wins
20:54.11Kobazyeah
20:54.29igcewielingI'm got it easy,  163 of my 164 endpoints have static IPs and auth by ip.
20:54.43igcewielingat least one one server.
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21:33.41vanczigcewieling: hm. thats a shame because the cheapest numbers seem to be in the us :I
21:52.03vanczigcewieling: can you give any pointers on what you said? "long code" seems to be sufficiently overloaded that im having trouble looking into it
21:53.17igcewielingvancz: it isn't a standard thing.  Almost no country supports tacking on random digits to phone numbers to create pseudo-DIDs.
21:53.40igcewielingCountries are moving away from variable length dialing, not towards it.
21:54.03vanczmmmm -_-
21:59.56igcewielingYou can dial all the extra digits you want, the usa telecos will only accept the expected number of digits.
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