IRC log for #asterisk on 20180214

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01:18.57*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.1 (2018/02/13), Standard: 15.2.1 (2018/02/13); 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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08:15.28elcontrastadorIs anyone using polycom vvx phones to receive multicast pages/streams? The paging settngs appear straightforward...the stream is def hitting the phones at the configured mcast addr/port.
08:16.36elcontrastadorI sending a raw stream to test here in the lab but intend to use MulicastRTP within asterisk as I do with the cisco sccp phones currently in asterisk with custom scripts
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08:19.16elcontrastadorlooking for the polycoms to just autoanswer on receipt of the mcast stream for use as an all call
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09:15.16Demon_VoIPasterisk 15.2.0 pjsip. 3-5 times per day asterisk hangs (taskprocessors "In Queue" for pjsip/default-* and netstat UDP socket queue growing). At hangs "core show locks" shows only one section. But allways the same. Why is it one? And what does it mean?
09:16.22Demon_VoIPhttps://pastebin.com/rUnaGutM
09:19.13wyoungDemon_VoIP: Lets VoIP
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09:19.54Demon_VoIPwyoung, Sorry, I don't understand.
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09:41.31wyoungDemon_VoIP: oh
09:41.56wyoungAre you from Sweden?
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09:52.54Demon_VoIPwyoung, is it sweden channel? I guess no. If you don't have a constructive answer to my question or suggestions for checking, then why trash?
10:12.53SamotUhm.
10:13.10SamotDemon_VoIP: So this happens 3-5 times a day, every day?
10:20.12Samot??
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10:22.32Demon_VoIPSamot, yes. And it's very bad for the production.
10:22.46SamotWhen did this start?
10:23.29Demon_VoIPNow using res_resolver_unbound.so. Without it even worse.
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10:24.23Demon_VoIPSamot, I can't say. Long ago. There's always something wrong. When one reason is finally overcome, others appear.
10:24.43SamotWell..
10:24.48Samot15 is only 4 months old.
10:24.53SamotSo did this only start on 15?
10:25.06SamotOr did you update to 15 and this was happening before..
10:25.22Demon_VoIPSamot, I was going to provide data, create another issue, but I can't interpret why there are no locks.
10:25.47filethat only shows you Asterisk locks, it doesn't show you PJSIP locks as well
10:25.49SamotIs this a new install or an upgrade?
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10:26.06fileit could very well be https://issues.asterisk.org/jira/browse/ASTERISK-26686
10:26.16Demon_VoIPSamot, Used 15.0, 15.1 and now 15.2 immediately after their release. Every time I hope that now everything will work.
10:26.44fileyou also use Asterisk at a scale that most people don't so it pushes things harder
10:27.10SamotOK so this is a fresh install with 15 ?
10:27.16Demon_VoIPfile, thanks. I watching this issue. I'll try workaround.
10:27.29SamotHow many users on the box?
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10:28.40Demon_VoIPfile, i have seen you rewrite keep_alive. will it be in new release?
10:28.49fileI haven't rewritten that.
10:29.05fileI changed OPTIONS support which is unrelated, and it is still up for code review
10:29.19SamotHow many users on the box?
10:30.13Demon_VoIPSamot, a lot of :( 3757 endpoints and 603 registrations, and a little motif, iax2
10:30.35SamotAnd what is the CPU/RAM of the system?
10:30.42Demon_VoIPfile, is where patch for 15 with OPTIONS rewritten?
10:30.47filethere is no patch for 15
10:31.05fileI'm not going to write support for all branches and then have to change each one as it goes through code review
10:31.06Demon_VoIPSamot, 4 CPU, 16 Gb ram.
10:31.11Samot.....
10:31.11fileit already took me almost 2 weeks to rewrite support
10:31.13SamotWhat?!
10:31.24SamotAre you serious?
10:31.59SamotThat is underpowered.
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10:32.19Demon_VoIPSamot, Load average: 1.44 1.39 1.38
10:32.40Demon_VoIPfile, thanks again. I'll wait with patient :(
10:32.59SamotYou need to scale this better.
10:33.08Demon_VoIPand with TLS trouble also :(
10:33.09SamotYou're running at almost 2 core
10:33.14SamotNo kidding.
10:33.31samwieremaRight now, in our Asterisk setup, the internal IP is used in the Contact header. How can I set it to use the public IP (using PSJIP, or maybe even ARI)?
10:33.44Demon_VoIPSamot, it's free public service. No pay, no garantee.
10:33.52SamotSetup your localnet and external networks properly.
10:34.00SamotDemon_VoIP: So?
10:34.07SamotYou're creating your own problems.
10:34.15SamotThis has nothing to do with who pays you.
10:34.45Demon_VoIPSamot, your suggestions?
10:34.58SamotEither seriously upgrade the resources on that box
10:35.07SamotOr don't run everyone on 1 box.
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10:35.55Demon_VoIPSamot, will bugs disappear?
10:36.11SamotYou mean will lockups stop?
10:36.14SamotProbably.
10:36.19SamotSince you wouldn't be dogging the system
10:37.27SamotAsterisk is not known for scalability
10:38.00fileit's not something anyone has focused on, as the people who truly want to push Asterisk that far on a single system aren't a lot when you can just scale out
10:38.13Samot^^^^^^
10:38.22SamotHell..
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10:38.33SamotEven BroadWorks, the ISP standard isn't _one_ system.
10:38.37SamotIt's five.
10:38.48SamotBecause the load is broken up based on it's job.
10:41.35SamotDemon_VoIP: If you step back and do the maths, would you put 1,000 users/endpoints on a 1 Core / 4 GB RAM system?
10:41.44SamotNo, because that would be insane.
10:42.44SamotEspecially with TLS connections.
10:43.25Demon_VoIPSamot, where are îãûå 1-10 simultaneous call. It does not load. The main load due to inefficient OPTIONS, for example.
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10:44.00SamotWhy do you mean "inefficient"?
10:44.07sekilhello
10:44.08Samots/Why/What/
10:44.41Demon_VoIPI fully agree that Asterisk is bad scales. But that is no reason to grow instances. This is a reason to change it to something else. Study, but not ready yet.
10:44.55SamotNo.
10:44.58SamotThat's not it at all.
10:45.17SamotYou have almost 4K endpoints not to mention trunks to peers/PSTN or whatever..
10:45.23SamotIf you're qualifying those..
10:45.29SamotThat's a lot of OPTIONs.
10:45.34SamotNot to mention REGISTERs
10:45.45SamotOr if the endpoints or other side is sending OPTIONs to you..
10:46.49SamotNot to mention MWI subscriptions, NOTIFYs, etc.
10:48.02SamotAnd what are  you going to change to?
10:48.13SamotFreeSWITCH? You're still in the same boat.
10:48.25Demon_VoIPmany years ago was SER, after that asterisk 1.2 (if I'm not mistaken). And now...
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10:48.40SamotSER is not and never has been a PBX
10:48.44SamotIt's a SIP Proxy
10:48.47Demon_VoIPFreeSWITCH with a lot of threads, Yate with worse support...
10:48.50SamotIt doesn't do _media_
10:49.32SamotYou need to either a) increase the CPU/RAM on this box or b) Roll out another system to move some of the load to.
10:49.49SamotJamming 4K + users plus trunks and other crap on a single box is bad.
10:49.54SamotAll around.
10:49.54Demon_VoIPSamot, I have a lot of php scripts via AGI, a lot of AMI handlers... Sometimes I think I do not care what PBX is behind all this
10:49.59SamotSpecially when that box is not right.
10:50.08SamotWell
10:50.14SamotSince AGI and AMI are _Asterisk_ things.
10:50.16SamotYou should.
10:51.29SamotAs file said, people who run this scale of users are just running a single Asterisk box.
10:51.40Samots/are/aren't/
10:51.56Demon_VoIPIf you do not want to offer money for new box, then the conversation can be over. If you wish, we are waiting for in private :) This is my hobby. I earn a lot of money, but I follow the principle: I spend only my time on Hobbies.
10:52.09SamotOK
10:52.16SamotSo let's be clear.
10:52.23SamotYou've asked for help on your issue.
10:52.39SamotYou were told the proper thing to do to help solve your issues
10:52.45SamotYou are refusing to follow that.
10:52.54SamotPart of your issues are your own making.
10:53.19SamotHow this is deployed is part of the source of your issues.
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13:50.41Demon_VoIPpjsip. Incoming call with codecs g722,alaw to endpoint A with codecs order alaw,g722. It changes the order of codec offeres to device. Is it possible to configure codec list (not order). or not to interfere in the order, no list of codecs. That the parties agreed on them without the server?
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13:53.32[TK]D-FenderNo.  * is a B2BUA and doesn't forward anything, it makes its own invites.
13:53.49[TK]D-FenderThe only time you'll get what you're looking for is in the case of a re-invite probably
13:55.50Demon_VoIPuse PJSIP_SEND_SESSION_REFRESH ?
13:57.18filehttps://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_MEDIA_OFFER
13:57.21filemight do it, dunno
13:57.26[TK]D-FenderNot sure about pjsip.  Haven't really touched it much myself
13:57.42[TK]D-FenderI need to catch up with hands-on experience
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14:03.22Demon_VoIPfile. read channel codecs from first device and write to PJSIP_MEDIA_OFFER(audio) ?
14:06.04filetry it and see
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17:14.00NonSecwitterare tags in From and To unique per phone / per transaction, per phone / per dialog, per phone / per call, or just per phone
17:14.03NonSecwitters/phone/ua
17:17.02igcewielingFrom and To?  I don't  think they are unique at all.
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17:19.49NonSecwitteri think it's per dialog, but i'm not 100% sure
17:21.00igcewielingyou could look at https://tools.ietf.org/html/rfc3261#page-50 and/or https://tools.ietf.org/html/rfc3261#section-19.3 but like may RFCs, it is confusing
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17:27.16SamotEach transaction store that information in memory
17:27.37SamotA dialog is basically an ongoing series of transactions.
17:28.34SamotSo if your phone sends both a REGISTER and an INVITE at the same time, for example, they are each their own transactions and their data from the SIP message is store based on the transaction.
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19:57.30lwlvltoday I made a very interesting discovery....I thought it'll be a good idea to test my network-performance...and whoops....my server is only connected with 100Mbit....but has a 1000Mbit nic inside. the switch is switched to auto/1000
19:57.37lwlvlany ideas? wrong driver?
19:59.33lwlvloh...wrong channel...
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21:36.27vthmm, with one phone, with SIP INVITE header To: <sip:065113XXXX@172.22.1.4;user=phone> I have the expected : Looking for 065113XXXX in International (domain 172.22.1.4). But with an incoming call from a trunk, with SIP INVITE header To: <sip:018364XXXX@10.7.1.65;user=phone>, Asterisk is Looking for s in in-ovh (domain 172.22.1.4) ...
21:36.52vtany idea why it is not looking for 018364XXXX in in-ovh ?
21:38.53vtthe sip.conf configuration of both lines seems pretty standard. I do remember having this problem when I was using pj_sip as well
21:46.05[TK]D-Fenderbecause "to:" is not the destination
21:46.24[TK]D-Fenderi's the INVITE BLAH@BLAH line at the start of the packet
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23:29.24Oooohboycan anyone tell me where to set the LUA path in asterisk? I see mention of it in the docs once, but no where else...
23:31.54Oooohboyaccording to [this](https://wiki.asterisk.org/wiki/display/AST/Lua+Dialplan+Tips+and+Tricks) I'm supposed to use the dotfile require method for breaking up files, but I need to set (or just know) the LUA_PATH value first
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