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| 00:19.41 | *** 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: 13.27.0 (2019/5/30) 16.4.0 (2019/5/30), 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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| 09:38.34 | forgotmynick | In res_odbc.conf can a hostname be specified? |
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| 12:53.00 | sibiria | i'm playing around with an anonymous endpoint in pjsip for some testing |
| 12:53.33 | sibiria | it does not send/accept registrations, but when various scanners try to register with it it still insists on logging a warning about an AOR not found for the endpoint |
| 12:54.02 | sibiria | is there a way to have pjsip *not* do this and simply ignore registrations as per the config for the endpoint? |
| 12:54.12 | file | no |
| 12:54.29 | sibiria | ok so it will always respond with a 404 and log the warning, got it |
| 12:56.16 | sibiria | something else i think i'm noticing looking at pjsip's log out is that it tries to send BYE repeatedly in hopes to get a 200 OK back? |
| 12:56.29 | sibiria | can pjsip's patience with this be tuned? |
| 12:56.52 | file | the timers configured in pjsip.conf control it, they are at a global scale and not per endpoint |
| 12:57.33 | file | https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L1008 |
| 12:57.44 | sibiria | looks like it's repeating BYE about 20 times before finally giving up on getting an OK back |
| 12:57.54 | file | https://andrewjprokop.wordpress.com/2013/07/02/understanding-sip-timers-part-i/ |
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| 12:58.27 | sibiria | thanks, file, i'll have a read |
| 12:59.47 | file | I'm probably thinking of the wrong timer though, it's always INVITE people are worried about and not BYE so you'll probably have to go look |
| 13:01.47 | file | looks like I answered this for someone last year, https://community.asterisk.org/t/session-termination/77659 and I don't think a timer_f option has been added since |
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| 13:07.11 | Samot | It's the retransmission timer. |
| 13:07.28 | ajayahmed | hi. i have an ip phone connected to asterisk extension and is a static member of a queue. when the handset rings but does not answer, the missed call does not appear on the handset. i understand this is the normal behaviour but was wondering if i can configure the handset to show the missed call? i'm not sure if this is something i need to configur |
| 13:07.29 | ajayahmed | e on the handset or the server so any pointers would be greatly appreciated |
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| 13:27.25 | sibiria | ah, will have to live with the retransmitted BYEs for now then. thanks file and Samot |
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| 13:28.49 | Samot | I guess the real question is, why are you not getting the 200 OK? |
| 13:30.11 | Samot | And if the other side hangs up, does Asterisk get the BYE and send the 200 OK? |
| 13:30.27 | Samot | Does that 200 OK reply just get sent once? |
| 13:31.01 | sibiria | botnet scanning |
| 13:31.17 | sibiria | conserving its traffic use i suppose |
| 13:31.42 | Samot | What do you need an anonymous endpoint for? |
| 13:32.17 | sibiria | nothing - this is just some tests i'm doing, and i got curious about the behavior |
| 13:35.37 | Samot | It's PJSIP's version of allowguests |
| 13:36.07 | sibiria | yeah that much i understand, it's the repeated BYEs that caught my eye, and i wondered if the retransmission was tunable |
| 13:40.56 | Samot | Well it would have nothing to do with a BYE. |
| 13:43.21 | Samot | I don't think PJSIP has that timer setting as an option yet. |
| 13:43.35 | sibiria | yeah it's just as file and you said |
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| 13:44.00 | sibiria | it has options to tune timer A and B, but not F which as far as i understand is the "BYE phase" |
| 13:44.17 | Samot | T1 and B |
| 13:44.27 | Samot | B is for INVITEs. |
| 13:44.31 | sibiria | t1 and b, yes, sorry |
| 13:46.21 | Samot | F isn't a settable timer |
| 13:46.27 | Samot | It's 64*T1 |
| 13:46.46 | Samot | So for PJSIP its 64*500 by default. |
| 13:47.13 | Samot | 32,000 |
| 13:54.02 | sibiria | something else i've noticed from a lot of these rogue clients is that they often compact headers. how compatible are those generally across the SIP ecosystem? |
| 13:54.23 | sibiria | works well or should preferably be avoided? |
| 13:57.20 | Samot | What do you mean "compact headers"? |
| 13:57.49 | sibiria | using the standardized single letter header name alternatives |
| 13:58.08 | sibiria | 'm' for contact, 'f' for from etc. |
| 13:58.59 | Samot | Show this.. |
| 13:59.39 | sibiria | this stuff: https://docs.oracle.com/cd/E17645_01/doc.50/e17654/sip_compact_form.htm |
| 14:00.26 | sibiria | i've never observed this with the commercial actors we work with, but the rogue clients and scanners seem to favor it |
| 14:01.36 | Samot | Well there is a reason for that. |
| 14:02.00 | Samot | Faster processing of the packets. |
| 14:02.09 | sibiria | yeah i can see the benefits. it's specified in the RFC but i suppose it's still uncommon, and perhaps not widely supported |
| 14:02.45 | Samot | Clearly it's supported by Asterisk. Otherwise it would complain the SIP packet was malformed. |
| 14:03.01 | sibiria | pjsip manages it fine, yeah. no idea about chan_sip |
| 14:03.11 | sibiria | pjsip has an option to also respond with compact format |
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| 17:31.25 | infinity_ | can anyone recommend a call reporting package for asterisk/freepbx? |
| 17:33.11 | Samot | What type of reporting? |
| 17:33.13 | Samot | Just calls? |
| 17:35.23 | infinity_ | incoming calls |
| 17:35.34 | infinity_ | and information about who answered, ring counts, stuff like that. |
| 17:35.52 | infinity_ | preferably not requiring queues, but maybe thats not possible |
| 17:37.51 | Samot | AsterNic CDR |
| 17:38.36 | infinity_ | I looked at that one and it seemed like the reports were pretty basic, but I can give it another look i suppose. |
| 17:40.33 | infinity_ | freepbx package $800 seemed a bit steep for what the video demo'ed. |
| 17:41.48 | mvanbaak | look at it this way: Can you do it in-house for less then $800? go for it. if not, pay up |
| 17:42.27 | infinity_ | I bet you could hire someone online to make that for about that. heh |
| 17:42.46 | mvanbaak | so, you have your answer there |
| 17:43.14 | infinity_ | it's almost like some queries + excel would be the way to go |
| 17:43.50 | mvanbaak | I can tell you $800 will be hardly enough to cover the costs to write up the technical specs of what you want to do |
| 17:44.46 | mvanbaak | purely made up numbers but easy to follow. a guy makes 50/hour, which means you pay 100/hour. so this single person, 1 hour of meeting about the tech implications == $100 |
| 17:44.55 | mvanbaak | that's how fast the money flows |
| 17:46.10 | mvanbaak | Asternic cdr is not cheap, but also the price is pretty fair for shops that dont have ppl with experience in cdr reporting and time |
| 17:47.11 | infinity_ | It seems there must be some legit reporting software for this. everything I've seen so far looks amature. I'd pay 2k for something that actually did some real reporting. |
| 17:51.02 | mvanbaak | I think most is in-house real specific software |
| 17:51.21 | mvanbaak | I created some in my career. all in-house and all very specific to the internals of the company |
| 17:52.10 | Samot | infinity_: Once Asterisk 13 hit a pattern formed. |
| 17:52.32 | mvanbaak | infinity_: Good luck with the search. hope you find something that fits your need |
| 17:52.37 | mvanbaak | for me it's game time |
| 17:52.40 | mvanbaak | later all |
| 17:52.46 | Samot | infinity_: Many third party companies/projects that did things for Asterisk either hit a really slow development stage or died. |
| 17:53.29 | Samot | Because 12/13 saw introductions of new things and overhauls of some existing things. |
| 17:55.00 | infinity_ | this one is free and might be the best i've seen: https://github.com/cdr-stats/cdr-stats |
| 17:55.13 | infinity_ | No idea what the status is though. |
| 17:56.15 | Samot | The last updates would be an indicator. |
| 17:57.06 | Samot | Oh yeah, that's a Star2Billing project. |
| 17:57.22 | Samot | They gave up the ghost on Asterisk stuff a while ago. |
| 17:57.42 | infinity_ | everyone is moving to the cloud :) |
| 17:58.19 | file | reporting and call billing are both things that are hard to make a generic offering out of, because they are inherently business logic |
| 17:59.02 | Samot | There is that. |
| 17:59.19 | Samot | However, if an app costs $800 to license... |
| 17:59.41 | Samot | Then you're not going to get the same caliber of an app just for $800. |
| 17:59.54 | Samot | Because it probably cost $15,000 to develop. |
| 17:59.54 | infinity_ | it seems that without something like SSRS and a lot of views that is the case. |
| 18:30.05 | forgotmynick | I have asterisk installed on a VPS and remote devices connect to extensions on this server. For the VPS firewall, aside from 5060, 5061, do I need to allow inbound traffic from the RTP port range? |
| 19:10.57 | [TK]D-Fender | If you want audio, yes |
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| 19:29.58 | sibiria | stateful firewall + symmetric RTP = you don't need to explicitly let traffic in for your specific scenario |
| 19:30.07 | sibiria | but it's a good practice anyway to avoid future problems |
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