| 00:00.38 | TurBoss | gn8 |
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| 00:49.02 | tengulre | hi, all |
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| 01:04.34 | kitsune | hi |
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| 01:18.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.4 (2017/12/13), Standard: 15.1.4 (2017/12/13); 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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| 12:28.26 | ashutoshkrchaube | Hi I am working on an asterisk server which is behind a router with port forwarding of port 5060. I am trying to register to the asterisk server via sip through the routers public ip. I can see the request coming to the server but its not coming to asterisk. any suggestions? |
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| 14:15.56 | bittis | coming to the server? how do you see it? |
| 14:16.22 | bittis | random question, disposition says busy, billsec greater than 0, which to trust? |
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| 15:02.33 | thiagoc | hi all, I have a PJSIP trunk who is dropping the calls after 3 minutes... on the logger I see that they send a BYE, what option should I send a "keepalive" packet to provider? |
| 15:04.25 | [TK]D-Fender | you shouldn't have to send a keep-alive |
| 15:04.39 | [TK]D-Fender | and jsut because they send a BYE doesn't explain why |
| 15:04.42 | [TK]D-Fender | and we aren't looking at the call |
| 15:13.11 | thiagoc | [TK]D-Fender, another provider disabled a "options parameter" on their side and it works, so I think there's some configuration missing on my side... when I was using chan_sip it doesn't occurs, so I think pjsip needs some parameter to keep the call alive |
| 15:13.58 | [TK]D-Fender | wrong |
| 15:14.26 | [TK]D-Fender | And we're still not looking at the call |
| 15:15.48 | [TK]D-Fender | * doesn't care about OPTIONS sent from the outside, and your system isn't the side that's deciding to end the call from what you said |
| 15:16.39 | thiagoc | ok, will try to get the logs |
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| 15:28.16 | kitsune | Report on the double NAT w/ AWS from last week: AWS appears to filter traffic based on source IP and port instead of just on source IP so it will block inbound traffic from a randomized port on the far end. |
| 15:31.43 | kitsune | So basically the RTP ports have to be opened on AWS. |
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| 16:04.32 | mvanbaak | kitsune: you can actually disable that |
| 16:05.39 | mvanbaak | kitsune: http://docs.aws.amazon.com/AmazonVPC/latest/UserGuide/VPC_NAT_Instance.html#EIP_Disable_SrcDestCheck |
| 16:06.44 | mvanbaak | it's enabled by default on all ec2 instances |
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| 16:58.15 | thiagoc | [TK]D-Fender, https://pastebin.com/raw/2F31hAJ0 |
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| 17:07.29 | [TK]D-Fender | <--- Transmitting SIP request (953 bytes) to UDP:179.124.44.234:5060 ---> [651/1910] |
| 17:07.29 | [TK]D-Fender | INVITE sip:047999712337@179.124.44.234 SIP/2.0 |
| 17:07.35 | [TK]D-Fender | You are sending a call over the internet |
| 17:07.57 | [TK]D-Fender | o=- 1230364827 1230364827 IN IP4 192.168.44.6 |
| 17:08.01 | [TK]D-Fender | And shouldn't be sending that IP |
| 17:09.11 | thiagoc | [TK]D-Fender, but the call works for about 3 minutes |
| 17:10.43 | [TK]D-Fender | Go ask your provider why |
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| 17:24.01 | mahafyi | hello, to read security events via an AMI script, what read value should be given for a manager.conf user? example read=system,verbose etc? |
| 17:24.54 | [TK]D-Fender | read the sameple config |
| 17:24.57 | [TK]D-Fender | they're all listed there |
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| 17:29.25 | mahafyi | [TK]D-Fender, thanks, i have it now |
| 17:29.38 | [TK]D-Fender | glad to hear |
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| 18:59.03 | AnonymooseWork | hey all |
| 19:00.59 | AnonymooseWork | Anyone up for a question about elastix / asterisk with a queue? |
| 19:01.30 | AnonymooseWork | Trying to figure out an easy/user friendly method for my helpdesk team to forward our helpdesk DID to a mobile on a schedule basis |
| 19:08.58 | [TK]D-Fender | So far those are 4 completely different things without a very clear line of when something is involved VS not |
| 19:09.36 | [TK]D-Fender | DID leads to a destination. a queue is a destination. "to a mobile" could be the end result of many different things |
| 19:10.06 | AnonymooseWork | Ok, |
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| 19:10.13 | AnonymooseWork | so (not a phone guy) but here was my idea |
| 19:10.21 | AnonymooseWork | DID >> Queue |
| 19:10.30 | AnonymooseWork | Helpdesk team dials into queue |
| 19:10.47 | AnonymooseWork | now for 'on call' the person on call would dial into queue and use follow me to forward to mobile? |
| 19:10.55 | AnonymooseWork | this could be the completely wrong approach |
| 19:11.30 | AnonymooseWork | My basic requirements are to ring an extension during office hours, and easily be able to forward to a mobile (the number will change depending who is on call) |
| 19:11.33 | [TK]D-Fender | CALLERS dial into a queue |
| 19:11.51 | AnonymooseWork | so an agent would... subscribe to the queue? |
| 19:11.58 | [TK]D-Fender | your extensions may be either fixed members, or ones that can can "login/out" |
| 19:12.17 | AnonymooseWork | our team travels quite a bit so being able to login/out would be benefiicial |
| 19:12.20 | [TK]D-Fender | "dynamic" |
| 19:12.24 | AnonymooseWork | that way people in the office could login |
| 19:12.24 | [TK]D-Fender | as the section says |
| 19:12.29 | AnonymooseWork | yes - i got that much :) |
| 19:13.07 | [TK]D-Fender | So let's say you have 2 people in the queue. should member 1 be ale to send their calls out? |
| 19:13.48 | AnonymooseWork | Well - What I was thinking was that agents would logout of the queue at the end of day |
| 19:13.51 | AnonymooseWork | except the agent on call |
| 19:14.00 | AnonymooseWork | then yes it would send their calls out |
| 19:14.17 | AnonymooseWork | Unless you can suggest a more suitable approach for this? |
| 19:14.37 | AnonymooseWork | I have two teams here that have on call rotations, but they are both small (one team is 3 people, the other is 5) |
| 19:15.14 | [TK]D-Fender | Are they on a fixed schedule? |
| 19:15.24 | AnonymooseWork | no |
| 19:15.45 | AnonymooseWork | We set the schedules on a weekly basis |
| 19:15.54 | AnonymooseWork | but it depends on who is available |
| 19:16.14 | AnonymooseWork | If teammembers are traveling for work they don't participate in the queue / on call that week |
| 19:16.26 | [TK]D-Fender | Followme may work. You'll need to enable call-confirmation because you don't want it hitting VM on the user's phone. |
| 19:16.46 | [TK]D-Fender | And a more sensible limit to not continuously dial them |
| 19:17.08 | AnonymooseWork | So I tested this; I did not have the confirmation on mind you |
| 19:17.12 | AnonymooseWork | but it didn't ring on my mobile |
| 19:17.16 | [TK]D-Fender | Or use a TC to check for your general hours and have that destination be a Virtual Extension whose FM you swap around |
| 19:17.18 | AnonymooseWork | it showed up as a missed call |
| 19:17.39 | [TK]D-Fender | it can't be a missed call if it didn't arrive at your phone |
| 19:17.46 | AnonymooseWork | true |
| 19:17.54 | AnonymooseWork | its almost as if it just hung up instantly though |
| 19:18.35 | [TK]D-Fender | probably not enough RING TIME |
| 19:18.41 | AnonymooseWork | on the follow me setup? |
| 19:18.49 | [TK]D-Fender | don't forget cell calls don't set up that fast compared to other forms. |
| 19:18.54 | [TK]D-Fender | on the QUEUE |
| 19:19.02 | [TK]D-Fender | remember if you're using the queue that's MEMEBER dial time |
| 19:19.10 | [TK]D-Fender | which isn't great |
| 19:19.16 | AnonymooseWork | oh ok |
| 19:19.31 | [TK]D-Fender | which is why it's better to use a TC to not let it hit the queue at all and use a dial method you can fix the time for |
| 19:19.32 | AnonymooseWork | yea - i think you are correct the agent timeout on the queue is set to 15seconds |
| 19:20.14 | AnonymooseWork | ok - so if i were to do that and have the time condition forward it to a virtual extension with FM set to 'on call mobile #1' |
| 19:20.23 | AnonymooseWork | is there an easy method you can think of the change the on call # |
| 19:20.47 | AnonymooseWork | i'd like to be able to program a button on their phone that says something like 'I'm on call' |
| 19:20.55 | mvanbaak | Last time I implemented something like that I used a database for that |
| 19:21.17 | AnonymooseWork | mvanbaak - hehe with the schedule in it? that would be brilliant |
| 19:21.25 | mvanbaak | yes |
| 19:22.14 | [TK]D-Fender | AnonymooseWork> is there an easy method you can think of the change the on call # <- use the feature code to change the Fm dest |
| 19:22.28 | AnonymooseWork | ah thx [TK]D-Fender |
| 19:22.52 | mvanbaak | or if you want to stick to a pure asterisk setup without a database and all, you can use .. yeah, what [TK]D-Fender said |
| 19:23.27 | AnonymooseWork | mvanbaak - i guess my prob with the db is that adds another layer of management to it |
| 19:23.34 | AnonymooseWork | and the team member on call could change |
| 19:23.37 | AnonymooseWork | IE family emergency |
| 19:23.37 | mvanbaak | yeah |
| 19:23.39 | AnonymooseWork | or something like that |
| 19:23.54 | AnonymooseWork | a button / code would be better |
| 19:23.57 | AnonymooseWork | i think |
| 19:24.12 | mvanbaak | we already had a schedule in a database for SMS/on-call duty so it was easy to hook asterisk to it |
| 19:24.27 | AnonymooseWork | makes sense |
| 19:24.34 | AnonymooseWork | so just to summarize |
| 19:25.06 | AnonymooseWork | err |
| 19:25.22 | AnonymooseWork | What do i associate the TC with? |
| 19:25.38 | [TK]D-Fender | TC -> workhours =- queues. Not work-hours = virtual extension w/ FM |
| 19:26.23 | [TK]D-Fender | or IIRC you might not have as much control over ring-time use a ring group instead with that same virtual extension |
| 19:27.41 | AnonymooseWork | ok |
| 19:28.42 | AnonymooseWork | Ok I understand that far |
| 19:28.47 | AnonymooseWork | as far as the TC |
| 19:28.55 | AnonymooseWork | What would callers call into? |
| 19:30.12 | [TK]D-Fender | caller = DID -> TC |
| 19:31.02 | AnonymooseWork | ok i'm looking at the inbound route for the DID |
| 19:31.08 | AnonymooseWork | and I don't see an y options for assigning a TC? |
| 19:31.52 | [TK]D-Fender | because you didn't CREATEone yet |
| 19:32.01 | AnonymooseWork | ok |
| 19:32.05 | [TK]D-Fender | You don't get drop-down options for things that don't exist yet |
| 19:32.30 | AnonymooseWork | ah |
| 19:32.31 | AnonymooseWork | thanks |
| 19:32.32 | AnonymooseWork | i feel dumb :) |
| 19:32.51 | AnonymooseWork | i've not worked with elastix much before |
| 19:39.47 | AnonymooseWork | thanks for your help |
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| 20:17.43 | startledmarmot | cresl1n: quick question - is there a relatively straightforward way to identify which SSRC is which registered pjsip endpoint when mixing in a confbridge? |
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| 20:18.03 | cresl1n | Ummm |
| 20:18.26 | startledmarmot | i.e. we need to identify a particular God-style user who needs certain placement on the UI in every browser |
| 20:18.44 | cresl1n | Great question |
| 20:18.47 | cresl1n | ponders |
| 20:19.28 | cresl1n | IIRC the SSRCs are declared in the SDP |
| 20:19.57 | cresl1n | That might be one option |
| 20:20.01 | cresl1n | kinda hacky |
| 20:20.15 | startledmarmot | Yeah, I thought so too |
| 20:20.32 | startledmarmot | I also didn't see that the SSRCs corollated across INVITEs, so it didn't seem like I could track it that way |
| 20:20.51 | startledmarmot | they seemed to be unique to that particular stream |
| 20:21.04 | cresl1n | Yeah, I don't think they are going to be universal across a mixing bridge |
| 20:21.18 | cresl1n | file probably know better |
| 20:21.19 | file | they are independent, and SSRC is only declared in WebRTC land |
| 20:21.46 | startledmarmot | makes sense. hm. |
| 20:21.48 | cresl1n | startledmarmot: I'm assuming you're using the new SFU code? |
| 20:21.53 | cresl1n | with browsers and such? |
| 20:22.01 | startledmarmot | you betcha |
| 20:22.08 | cresl1n | That's pretty awesome! |
| 20:22.37 | cresl1n | I'm trying to think of a good way to do it |
| 20:22.59 | cresl1n | So you're trying to figure out which browser is this "special" sip endpoing |
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| 20:23.03 | cresl1n | *endpoint |
| 20:23.13 | cresl1n | and if you know the endpoint name, you know who it is? |
| 20:23.29 | cresl1n | Actually, the browser should know which one he's registered as, no? |
| 20:23.30 | startledmarmot | Everything works A+ so far, btw - it's the other integration partner that is having all the issues |
| 20:23.36 | startledmarmot | But yes, that's correct |
| 20:24.15 | startledmarmot | We have a particular "special" user that needs to be identified on each browser screen |
| 20:24.27 | cresl1n | Oh, like their video stream is special |
| 20:24.29 | startledmarmot | so they can have their video element placed in a certain location |
| 20:24.29 | cresl1n | for all the users? |
| 20:24.31 | startledmarmot | yeah, exactly |
| 20:24.35 | cresl1n | GOTCHA! |
| 20:24.37 | file | we don't really exchange that kind of information... the best you could do easily would be to put the name of the stream in the SDP for that stream, get it out of the SDP in the browser, parse it, and do different placement |
| 20:24.44 | cresl1n | I don't know if there's a clever way to do that yet |
| 20:24.47 | cresl1n | but that'd be very useful |
| 20:25.03 | file | the name of the stream internally is the channel and a bit extra, which would include the endpoint name |
| 20:25.32 | file | otherwise the right way would be having a communication layer so you know who is each stream |
| 20:25.38 | file | and can treat it differently |
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| 20:25.51 | file | SIP MESSAGE requests, for example |
| 20:26.53 | startledmarmot | can the SDP be modified for a confbridge invite? |
| 20:27.07 | startledmarmot | the SIP MESSAGE idea is interesting |
| 20:27.24 | file | there's nothing built in to do this, you'd have to modify res_pjsip_sdp_rtp / res_pjsip_session |
| 20:27.31 | startledmarmot | gotcha |
| 20:27.40 | file | it only knows that it was told to reinvite with a specific set of streams |
| 20:27.48 | cresl1n | did gtjoseph_ ever get his SIP message patches merged? |
| 20:27.52 | startledmarmot | My first attempt was to add an X-Header that persisted into the ConfBridge that could just send the endpoint name |
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| 20:28.04 | file | cresl1n: mayyyyyyybe? |
| 20:28.08 | startledmarmot | hahaha |
| 20:28.26 | startledmarmot | loves the sound of a good mayyyyyyybe |
| 20:28.35 | gtjoseph_ | nope |
| 20:28.43 | gtjoseph_ | was going to get it done over the christmas wek |
| 20:28.59 | cresl1n | I'm trying to think if that would solve your problem though |
| 20:29.42 | cresl1n | it's almost like you need a way to query the local stream mapping to channel names or channel ids for the bridge |
| 20:30.04 | cresl1n | does that sound right startledmarmot? |
| 20:30.15 | startledmarmot | that would certainly do it |
| 20:30.24 | startledmarmot | although I'd even just take endpoint name of stream originator |
| 20:30.36 | startledmarmot | since that pjsip endpoint can be different from all the others |
| 20:31.01 | cresl1n | so if it's had the channel name to SSRC map, that'd work? |
| 20:31.03 | cresl1n | *it |
| 20:31.15 | cresl1n | channel name should include PJSIP endpoint id, no? |
| 20:31.28 | gtjoseph_ | well, my messaging patch does allow any text to be added to a channel and have it propogate to all clients as a SIP MESSAGE |
| 20:31.30 | startledmarmot | yes, that'd certainly do it |
| 20:32.49 | cresl1n | gtjoseph_: I'm trying to figure out if that would that solve his stream correlation problem though? |
| 20:33.17 | cresl1n | 'cause it sounds like he need to be able to identify which steam is a particular remote channel from the client side |
| 20:33.18 | gtjoseph_ | probably not. might still need a modification to create the text |
| 20:33.45 | gtjoseph_ | well, he could craft a MESSAGE with that info in it. |
| 20:34.06 | cresl1n | How would that help though, since SSRCs are not shared between bridge participants |
| 20:34.28 | cresl1n | I wonder if the stream names are |
| 20:34.34 | cresl1n | goes back to re-read what file said |
| 20:34.52 | gtjoseph_ | oh i see what you mean |
| 20:34.55 | cresl1n | I think he already answered this |
| 20:35.48 | cresl1n | So according to file, it might already be there? |
| 20:35.54 | cresl1n | in the mslabel maybe? |
| 20:36.06 | cresl1n | reads some more |
| 20:37.54 | startledmarmot | looks at his captures for mslabel |
| 20:39.12 | file | no |
| 20:39.15 | file | we don't put it into the SDP |
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| 20:39.32 | file | I was saying that the data is there internally, but you'd have to communicate it somehow |
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