| 00:20.08 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:20.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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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| 00:31.23 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
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| 00:53.34 | CRCinAU_ | so rather vague question - but where does pjsip fit in the development plans / goals / aims of Asterisk? |
| 00:53.37 | CRCinAU_ | I see 14.4 has a version of pjsip in with it |
| 00:53.55 | Samot | It's had a version since 12 |
| 00:54.07 | CRCinAU_ | but from having a tinker, it seems the config for pjsip is just..... aweful. |
| 00:54.13 | Samot | It's the primary SIP driver, as of now. |
| 00:54.36 | CRCinAU_ | wow, ok. |
| 00:54.48 | Samot | Chan_SIP = legacy |
| 00:55.14 | CRCinAU_ | see, I tried to set up just a single trunk - but couldn't get the inbound calling working at all |
| 00:55.18 | Samot | IIRC it is no longer being developed on and Asterisk development is based on PJSIP as the main driver. |
| 00:55.21 | CRCinAU_ | just that it couldn't find a peer |
| 00:55.39 | CRCinAU_ | sorry - endpoint. |
| 00:55.43 | CRCinAU_ | (new terms) |
| 00:55.58 | Samot | Old term |
| 00:56.02 | Samot | endpoint is the phone. |
| 00:56.12 | Samot | However, Chan_SIP only supported one endpoint |
| 00:56.18 | Samot | PJSIP does not. |
| 00:56.34 | Samot | It supports multiple endpoints on one user account. |
| 00:56.38 | CRCinAU_ | See, I have 5 total SIP trunks.... 2 with provider A, 2 with B, 1 on its own. |
| 00:57.09 | Samot | So with Chan_SIP would you need a trunk(peer) for each IP |
| 00:57.09 | CRCinAU_ | I register with a /$ext on the end of the chan_sip register line |
| 00:57.20 | CRCinAU_ | well, I have one for each account. |
| 00:57.27 | Samot | PJSIP can have multiple hosts with a peer |
| 00:57.46 | CRCinAU_ | with different username / password for each account? |
| 00:57.53 | Samot | No. |
| 00:58.18 | Samot | If you have actual SIP accounts you need to register to, you need a trunk per sip account. |
| 00:59.01 | CRCinAU_ | so the problem I think I was seeing.... I have context 'incoming' that all SIP trunks go to ... that has an extension that matches the /$ext and the end of each register line in chan_sip |
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| 00:59.30 | CRCinAU_ | What I think the problem was for incoming stuff is that it didn't have that extra info - so it couldn't find out where to send the incoming call |
| 00:59.53 | CRCinAU_ | .... and at 3am I just reverted everything and went to sleep ;) |
| 01:00.46 | CRCinAU_ | so is there a new way of doing that with pjsip? or is that unsupported? |
| 01:01.20 | Samot | I would have to look.. |
| 01:01.29 | Samot | I haven't moved to PJSIP at all. |
| 01:01.38 | CRCinAU_ | I just thought I'd give it a try |
| 01:02.08 | CRCinAU_ | I use sccp for the actual phones (apart from 2), so don't really care about the phone config side as much |
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| 07:27.32 | dokma | How to tell if noise comes from ATA || * || VOIP Phone ?? |
| 07:39.04 | dokma | I have a SPA942 VOIP Phone + a few Linphone instances <-> * <-> ATA <-> PSTN setup. |
| 07:39.44 | dokma | I've just made a Linphone to SPA942 call and when I use a headset there is no noise but without headset I hear a hiss. |
| 07:40.13 | dokma | I get that same hiss when a call is made from PSTN to SPA942 so I suspect it comes from the PSTN? |
| 07:40.25 | dokma | But I have no clue how to get rid of that... |
| 07:45.21 | dokma | I've used the headset on the Linphone side. On the SPA942 side I listened on the handset. The hiss is there even when there is no conversation at all. |
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| 08:04.02 | dokma | I'm trying to reduce volume gain on * and see if that has anything to do with it... |
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| 08:32.45 | TurBoss | Hi |
| 08:33.14 | TurBoss | Im trying to register to my isp sip but i'm getting Fatal 404 |
| 08:33.28 | TurBoss | res_pjsip_outbound_registration.c:909 handle_registration_response: Fatal response '404' received from |
| 08:41.45 | TandyUK | sounbds like you have the login name wrong |
| 08:41.55 | TandyUK | its not found, rather than wrong password |
| 08:57.52 | TurBoss | thanks will try |
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| 09:31.55 | CRCinAU_ | so |
| 09:32.00 | CRCinAU_ | gack to my questions about pjsip |
| 09:32.20 | CRCinAU_ | so I use the register => blah/123456 on my current chan_sip config |
| 09:32.45 | CRCinAU_ | then in the [incoming] context, I have extension 123456 which the incoming call goes to |
| 09:32.53 | CRCinAU_ | pjsip doesn't really seem to have anything comparable. |
| 09:32.59 | CRCinAU_ | so, how can I route the incoming calls? |
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| 09:52.46 | file | The contact_user in a registration section is the equivalent. |
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| 10:00.49 | CRCinAU_ | hrrrm ok |
| 10:00.57 | CRCinAU_ | might have another go at it then |
| 10:02.12 | CRCinAU_ | now the other thing... I have 5 lines between 3 sip providers. |
| 10:02.26 | CRCinAU_ | can I simplify the ones where server etc is the same somehow? |
| 10:02.38 | CRCinAU_ | ie not have 4 x config entries x 5 'lines'? |
| 10:02.52 | file | if it's 5 separate registrations, then it's 5 separate registrations |
| 10:03.11 | file | the only thing you can improve is use config templates to reduce the amount of stuff in the .conf file |
| 10:03.51 | CRCinAU_ | yeah ok... |
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| 11:53.34 | CRCinAU_ | file: so another question... Do I need an aor entry? |
| 11:54.12 | file | If you want to place outbound calls, yes. |
| 11:54.17 | CRCinAU_ | ie what I want to do is register, so I have a type = registration |
| 11:54.28 | CRCinAU_ | then I have another with type = auth |
| 11:54.43 | CRCinAU_ | so the example I see only has max_contacts = 1 |
| 11:54.57 | CRCinAU_ | it says 'Maximum number of contacts that can bind to an AoR'. |
| 11:55.07 | file | An AOR is unrelated to an outbound registration |
| 11:55.12 | CRCinAU_ | Does that equal the number of - say outbound simultanious calls? |
| 11:55.18 | file | No. |
| 11:55.34 | file | A contact is the address for how to contact something. |
| 11:55.46 | CRCinAU_ | so the example says: |
| 11:55.46 | CRCinAU_ | ;[7000] |
| 11:55.47 | CRCinAU_ | ;type=aor |
| 11:55.47 | CRCinAU_ | ;max_contacts=1 |
| 11:55.50 | file | For inbound registrations we can allow multiple. |
| 11:55.52 | CRCinAU_ | ... and that's it. |
| 11:56.41 | file | Yes. It limits the inbound registrations such that only one address is present. |
| 11:56.53 | CRCinAU_ | hmmm on |
| 11:56.57 | CRCinAU_ | ok even |
| 11:57.27 | CRCinAU_ | so as long as I set contact_user in the type=registration, that will let me use the same as /whatever at the end of a register line in the incoming context |
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| 11:57.48 | CRCinAU_ | (kind of a steep learning curve on this vs good old chan_sip format |
| 11:58.21 | file | Yes. |
| 11:58.34 | file | We have guides and such on the wiki and a wizard system to make it easier |
| 11:58.51 | CRCinAU_ | yeah - I tried the contrib script to convert my sip.conf to pjsip.conf... |
| 11:59.02 | CRCinAU_ | I think it seemed confused between the lines on the same provider |
| 11:59.06 | CRCinAU_ | ie with just a different username |
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| 11:59.48 | file | It's possible |
| 12:00.48 | CRCinAU_ | So, to make this easier.... if I have say, [7000] with just type = aor and max_contacts = 1, I can template that for each line.... |
| 12:01.00 | CRCinAU_ | is there a document somewhere on the templating? |
| 12:01.12 | CRCinAU_ | I'm using this at the moment: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 12:01.21 | file | As I said, that is only for outgoing. |
| 12:01.21 | file | Gah cellular. |
| 12:01.33 | file | An AOR has nothing to do with outbound registration. Only for outgoing calls and inbound registrations. |
| 12:01.57 | file | Templates are a config thing, they aren't specific to PJSIP |
| 12:01.59 | CRCinAU_ | *nods* but that means I'm going to have to have one for every 'line' as I have 5 different phone numbers |
| 12:02.04 | file | They'd be mentioned elsewhere |
| 12:02.18 | file | If they all go to the same place you can have one |
| 12:02.36 | file | An AOR can be referenced by more than one endpoint |
| 12:02.54 | CRCinAU_ | Hrrrrm |
| 12:03.25 | file | An AOR is where to contact something, an endpoint defines the configuration to use for contacting them |
| 12:03.35 | CRCinAU_ | ahhh - using the aors = [seomthing] in the endpoint. |
| 12:03.56 | file | aors=something |
| 12:04.07 | CRCinAU_ | so say, [aor]\ntype = aor\nmax_clients = 1 |
| 12:04.13 | CRCinAU_ | then in the endpoint: aors = aor |
| 12:04.16 | CRCinAU_ | that makes sense. |
| 12:04.57 | file | If you aren't doing an inbound registration you don't have to specify max contacts |
| 12:05.16 | file | And a contact entry in an AOR defines the SIP URI for contacting |
| 12:05.54 | CRCinAU_ | so do I need an endpoint for a SIP 'trunk'? |
| 12:06.33 | file | Yes. |
| 12:06.39 | CRCinAU_ | ie it will do both incoming and outgoing calls - ie via Dial(PJSIP something) |
| 12:06.57 | file | An endpoint defines the configuration to use. Auth, callerid, codecs |
| 12:07.45 | CRCinAU_ | ok - so I'm confused now.... |
| 12:07.46 | CRCinAU_ | https://github.com/mojolingo/asterisk/blob/master/configs/pjsip.conf.sample#L219 |
| 12:08.03 | CRCinAU_ | Line 291 says "endpoint configured as a trunk, inbound auth and registration" |
| 12:08.19 | CRCinAU_ | it has an endpoint, auth, aor .. but no registration |
| 12:08.25 | CRCinAU_ | ie no block for registration |
| 12:09.03 | file | Read the description below |
| 12:09.08 | file | That is for inbound registration |
| 12:09.16 | CRCinAU_ | ahhhh fek. |
| 12:09.37 | CRCinAU_ | yeah - ok... overview on my part. |
| 12:10.14 | file | The wiki has an itsp example, but I'm on a phone so finding it would prove difficult |
| 12:10.57 | CRCinAU_ | yeah - I think that's here: https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples |
| 12:11.05 | file | Probably |
| 12:11.09 | CRCinAU_ | I've been trying to cross-reference due to being a complete pjsip noob |
| 12:11.31 | CRCinAU_ | that has a registration, auth, aor and endpoint |
| 12:11.32 | file | Yup |
| 12:11.48 | file | And a match, to associate incoming requests |
| 12:11.51 | CRCinAU_ | I don't quite get what this is for in the aor though: contact=sip:203.0.113.1:5060 |
| 12:12.18 | file | That specifies how to contact it |
| 12:12.42 | file | That is a SIP URI where the itsp can be reached |
| 12:13.06 | CRCinAU_ | so that should be the same as the server_uri in the registration? |
| 12:13.21 | file | Yes |
| 12:13.37 | file | If they expect you to call the same place you register to, which most do |
| 12:13.40 | CRCinAU_ | ok - cos the example shows: |
| 12:13.41 | CRCinAU_ | server_uri=sip:sip.example.com |
| 12:13.43 | CRCinAU_ | but: |
| 12:13.47 | CRCinAU_ | contact=sip:203.0.113.1:5060 |
| 12:13.51 | CRCinAU_ | in the aor entry |
| 12:14.05 | file | Hostname or IP works |
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| 12:14.24 | CRCinAU_ | yeah - its just a little confusing as it seems like they're not related :) |
| 12:14.42 | file | If you comment someone will adjust |
| 12:17.58 | CRCinAU_ | comment added. |
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| 13:46.01 | CRCinAU_ | hrrrm |
| 13:46.25 | CRCinAU_ | so I've done something..... and registrations to 3 of SIP providers fails now with a 401. |
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| 14:04.48 | CRCinAU_ | that's interesting |
| 14:04.49 | CRCinAU_ | > pjsip list auths |
| 14:04.49 | CRCinAU_ | No objects found. |
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| 14:05.45 | CRCinAU_ | is there a problem having an auth object with the same name as an endpoint / aor ? |
| 14:05.49 | CRCinAU_ | as long as type = auth |
| 14:06.45 | file | Nope. There isn't. |
| 14:06.46 | file | If the config is wrong for an auth it won't load though and will tell you why at startup. |
| 14:09.39 | CRCinAU_ | hrrrm |
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| 14:10.36 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:10.51 | CRCinAU_ | ah ha! |
| 14:11.15 | CRCinAU_ | Config file 'pjsip.conf' could not be loaded; configuration contains a duplicate object: '901' of type 'auth' |
| 14:11.30 | CRCinAU_ | so if there's a single error, it fails to do everything |
| 14:12.02 | file | depends on the error |
| 14:12.03 | CRCinAU_ | ie every auth object failed. |
| 14:12.06 | CRCinAU_ | interesting to know. |
| 14:12.34 | CRCinAU_ | I typo'ed doing a copy / paste of 901 -> 902 and forgot to change the name |
| 14:17.16 | CRCinAU_ | you know, I think I'm almost there now...... |
| 14:17.18 | CRCinAU_ | *almost* |
| 14:18.10 | CRCinAU_ | all my SIP trunks register ok |
| 14:20.29 | CRCinAU_ | now I'm down to figuring out the Dial |
| 14:25.02 | CRCinAU_ | yay - first outbound call works :D |
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| 14:32.24 | CRCinAU_ | so now on incoming calls, I see this: Request 'INVITE' from '"$number" <sip:$number@$ip>' failed for '$ip:5060' (callid: 403b-154e-3112017143045-bn01-nme1-0-$otherip) - No matching endpoint found |
| 14:32.47 | CRCinAU_ | so it seems maybe the contact_user in the registration isn't doing its thing? |
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| 14:40.16 | CRCinAU_ | file: this is what I was talking about with the /blah on the end of the register line in sip.conf |
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| 14:41.00 | [TK]D-Fender | Tha has nothing to do with a contact user |
| 14:41.29 | [TK]D-Fender | because that would be the exten targeted by the INVITE, but in this case it's not what is being dialed, it's that it can't match the caller |
| 14:43.01 | CRCinAU_ | can't match the caller? You mean map the INVITE to an ENDPOINT? |
| 14:43.11 | [TK]D-Fender | yes, which is what it's saying |
| 14:43.23 | [TK]D-Fender | "I don't know you, BYE" |
| 14:43.25 | CRCinAU_ | but doesn't the endpoint name get set in the [xxxxxx] part? |
| 14:43.40 | [TK]D-Fender | That includes auth requirements, ACL's, etc |
| 14:43.48 | [TK]D-Fender | there are username on purpose too |
| 14:43.54 | [TK]D-Fender | don't assum that [] carries |
| 14:43.58 | [TK]D-Fender | just like chan_sip |
| 14:44.05 | [TK]D-Fender | [john] |
| 14:44.09 | [TK]D-Fender | defaultuser=fred |
| 14:44.09 | file | you can match an incoming call based on IP address using a match section |
| 14:44.13 | [TK]D-Fender | it's FRED. |
| 14:44.20 | [TK]D-Fender | Or by IP, etc |
| 14:44.22 | file | the "line" option in an outbound registration can also work for SOME providers |
| 14:44.26 | [TK]D-Fender | depending on the ID method |
| 14:44.40 | CRCinAU_ | file: yeah - but as I have different numbers that register to the same IP, I can't use that :\ |
| 14:45.07 | file | have them all go to the same context and then direct according to extension? |
| 14:45.15 | file | it's the same behavior as chan_sip |
| 14:45.39 | CRCinAU_ | well, they do all go to the same context. |
| 14:45.49 | CRCinAU_ | ie everthing is context = incoming |
| 14:47.35 | *** join/#asterisk theodotos (~theodotos@78.158.142.251) |
| 14:47.37 | CRCinAU_ | I'm not quite sure I understand what you propose.... |
| 14:47.59 | CRCinAU_ | so - you're saying match by IP to an endpoint, then use the extension in the context incoming to spit it somewhere? |
| 14:48.05 | file | sure. |
| 14:48.17 | file | that's not unique to chan_pjsip either - that's how it works in chan_sip |
| 14:48.31 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-sbfbygwuhlhyihiy) |
| 14:48.32 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:48.51 | *** join/#asterisk shido6 (~voipmonk@2604:a880:2:d0::191:8001) |
| 14:49.32 | CRCinAU_ | or would it be better to do it by user? Or will I know the user in that case? |
| 14:49.57 | CRCinAU_ | I'm just reading the Identify by User section of the PJSIP troubleshooting |
| 14:49.59 | file | it is unlikely your ITSP would do that. |
| 14:50.18 | CRCinAU_ | hrrrm |
| 14:50.27 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 14:50.37 | file | which is why when it comes to ITSPs people do IP based matching |
| 14:50.39 | file | OR you can try line |
| 14:50.54 | file | http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ |
| 14:51.11 | CRCinAU_ | haha yeah, just found that to read. |
| 14:52.47 | CRCinAU_ | hrrrm - that may well be what I was already using |
| 14:52.57 | CRCinAU_ | ie via chan_sip? |
| 14:53.03 | igcewieling | Thankfully chan_sip has far better documentation and community knowledge and chan_pjsip. |
| 14:53.26 | file | chan_sip doesn't support it |
| 14:53.55 | CRCinAU_ | hrrrrm |
| 14:54.14 | file | you were either doing anonymous or IP based matching |
| 14:54.20 | CRCinAU_ | igcewieling: I hear you.... this feels like discovering fire ;) |
| 14:54.26 | CRCinAU_ | interesting. |
| 14:55.06 | CRCinAU_ | so for completeness.... the contact_user in the registration is there to set the extension used in context incoming. |
| 14:55.31 | igcewieling | CRCinAU_: Asterisk's auth is a pain in the ass. I nearly disable Asterisk's auth (allowguest=yes) so I can handle it properly in the dialplan and do things like subnet matches. I don't recommend that for n00bs, it is far, far too easy to let unauthorized people dial out. |
| 14:55.34 | CRCinAU_ | and in theory, the line is there to try and pin an endpoint to the invite. |
| 14:56.09 | CRCinAU_ | igcewieling: that's an interesting idea...... but dangerous :P |
| 14:56.28 | igcewieling | CRCinAU_: I've been using Asterisk almost every day since 2001. |
| 14:57.25 | file | chan_pjsip allows subnet matching. |
| 14:57.46 | igcewieling | file: apparently not in the way we want to use it. |
| 14:58.10 | file | CRCinAU_: some ITSPs don't send the dialed extension in the incoming call, contact_user (and the /<extension> in chan_sip) is there in case they don't |
| 14:58.21 | igcewieling | host= is what seems to match, not permit=1.2.3.0/24 |
| 14:58.34 | CRCinAU_ | hrrrm - let me try it with line then and see what happens. |
| 14:58.47 | file | for chan_sip that is correct |
| 14:58.50 | CRCinAU_ | igcewieling: I've been banned from here under my normal nick since about then ;) LOL |
| 14:59.17 | file | chan_pjsip has a separate IP matching configuration which allows explicit hosts/IPs/subnets (and multiple of them) |
| 14:59.34 | CRCinAU_ | I had a disagreement about the number of versions of asterisk that were happening... making it a pain for 3rd party developers |
| 14:59.37 | file | it also gets all A records, and does SRV |
| 14:59.45 | CRCinAU_ | still a bit of a pain, but hey - people have dealt with it now. |
| 15:00.00 | *** join/#asterisk matrix1233 (~matrix123@197.2.214.155) |
| 15:00.18 | CRCinAU_ | so yeah, if anyone feels generous about unbanning my normal nick, that'd be great ;) |
| 15:00.28 | CRCinAU_ | but that's not really a pjsip thing..... |
| 15:01.09 | igcewieling | CRCinAU_: if they didn't want me here then I had no interest in being here. no big deal. someone removed the ban eventually. |
| 15:01.09 | *** join/#asterisk yokel_ (~yokel@unaffiliated/contempt) |
| 15:01.09 | igcewieling | you're not....._bkw, are you? |
| 15:01.15 | CRCinAU_ | nah - just CRCinAU |
| 15:01.46 | file | I don't see a ban for that. |
| 15:01.46 | CRCinAU_ | its a pain to change my nick each time I come in here. lol |
| 15:01.55 | igcewieling | file: I look forward to being able to rip out stuff I created to work around those chan_sip limitations. |
| 15:02.06 | *** join/#asterisk aiksa[LV] (5f822365@gateway/web/freenode/ip.95.130.35.101) |
| 15:02.28 | CRCinAU_ | file: it lets me join, but I can't type anything - so maybe its a mute? or something like that? |
| 15:02.38 | CRCinAU_ | irssi just says I can't send to channel - banned. |
| 15:02.40 | igcewieling | I might even be able to get rid of my AEL/SRV abomination I use to handle SRV currently. |
| 15:02.52 | file | I... don't know how to see that from here |
| 15:03.02 | file | igcewieling: we also do NAPTR/SRV with failover |
| 15:03.50 | CRCinAU_ | file: probably something like: /msg chanserv unquiet #asterisk crcinau |
| 15:04.27 | file | nope. |
| 15:05.32 | CRCinAU_ | I can't seem to see any option to even list the 'quiet'ed users |
| 15:05.38 | file | indeed |
| 15:06.42 | CRCinAU_ | except its gotta be on my nickname |
| 15:06.53 | CRCinAU_ | otherwise I couldn't talk under this nickname |
| 15:08.14 | aiksa[LV] | @file, Hi! Maybe you could look if I have submitted crash report correctly? |
| 15:08.39 | aiksa[LV] | https://issues.asterisk.org/jira/browse/ASTERISK-26935 |
| 15:08.42 | CRCinAU_ | let me change back to my normal username..... |
| 15:08.43 | *** part/#asterisk CRCinAU_ (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 15:08.47 | file | I did, I didn't have any comments |
| 15:08.51 | *** join/#asterisk CRCinAU (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 15:08.51 | file | and it'll be triaged more in the future |
| 15:09.37 | aiksa[LV] | Ok; because I can't seem to be able to create a minimal version where this could be reproduced |
| 15:11.27 | *** mode/#asterisk [-q crcinau!*@*] by file |
| 15:11.43 | CRCinAU | test |
| 15:11.47 | CRCinAU | eyyyyyyy |
| 15:12.08 | file | you were quieted but I had to guess |
| 15:12.17 | CRCinAU | LOL |
| 15:12.21 | CRCinAU | figures, nickname match. |
| 15:13.05 | CRCinAU | so. many. years. I've had to change the nickname :P |
| 15:13.15 | CRCinAU | file: thanks. |
| 15:13.22 | file | aye |
| 15:13.33 | CRCinAU | now to test this line bit |
| 15:15.09 | CRCinAU | interesting. |
| 15:15.15 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 15:15.20 | CRCinAU | I note I get: No matching endpoint found x 2, then Failed to authenticate |
| 15:15.24 | CRCinAU | then it dies. |
| 15:15.32 | CRCinAU | does this mean it attempts to use auth? |
| 15:15.41 | CRCinAU | or just 3 strikes and you're out |
| 15:16.13 | file | we challenge regardless |
| 15:16.30 | file | so that the behavior isn't different between an endpoint being found and an endpoint not being found |
| 15:16.30 | CRCinAU | m'kay.... so pretty much I have to do the IP matching.... |
| 15:16.39 | file | otherwise someone could enumerate and find endpoints |
| 15:16.44 | CRCinAU | true |
| 15:18.09 | igcewieling | <rant>*scream* I hate our ticketing system (Kayko)</rant> |
| 15:18.27 | CRCinAU | so as its missing from the documentation, can match be a hostname? |
| 15:18.37 | CRCinAU | or is it ONLY ip address? |
| 15:18.54 | CRCinAU | ie it doesn't do an A record lookup or anything |
| 15:23.23 | file | it can be a hostname, IP address, and include a subnet mask |
| 15:23.31 | file | and you can have multiple match= lines. |
| 15:27.23 | CRCinAU | ok, lets see how that goes. |
| 15:33.46 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 15:38.09 | CRCinAU | yay! |
| 15:39.10 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 15:40.02 | CRCinAU | that seems to be actually operational now :) |
| 15:40.22 | CRCinAU | thanks a ton file .... I wish it was a somewhat smoother learning curve... but that's rough :P |
| 15:40.43 | file | boom |
| 15:41.00 | CRCinAU | I've tested both incoming and outgoing (to only one trunk atm, but hey) |
| 15:41.16 | CRCinAU | I'll test more when people aren't asleep and I can go on a dial frenzy |
| 15:41.52 | *** join/#asterisk defswork (~aporter@212.38.66.116) |
| 15:41.55 | CRCinAU | at the moment, I put an identify block on each SIP trunk |
| 15:42.11 | file | if it's the same IP address you can't |
| 15:42.13 | CRCinAU | I probably don't need that - and could get away with one identify per provider (going from 5 -> 3) |
| 15:42.21 | *** join/#asterisk Samael28 (~Samael28@80.78.45.158) |
| 15:42.23 | file | others become noops |
| 15:42.34 | *** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 15:42.39 | defswork | I have log file entries like these - [Apr 11 16:12:11] NOTICE[12182][C-000003dd] chan_sip.c: Failed to authenticate device 201<sip:201@211.131.111.91>;tag=d91ecfbd - how can I get asterisk to log the source IP so I can block it (via fail2ban) ? |
| 15:42.44 | *** part/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 15:43.06 | CRCinAU | but as long as it gets punted into the right context, it'd just get the extension from the contact_user? |
| 15:43.16 | CRCinAU | defswork: log security level |
| 15:43.17 | file | yes. |
| 15:43.41 | CRCinAU | you know, that's probably the only reason my chan_sip setup actually worked for all lines then |
| 15:43.49 | CRCinAU | because I have the same context for all incoming numbers. |
| 15:44.22 | CRCinAU | defswork: and be prepared to filter through a TON of crap |
| 15:45.38 | CRCinAU | file: I figure if I come back to this in a few years time, having an identify block for each SIP trunk will make more sense to me ;) |
| 16:10.12 | *** join/#asterisk s-mutin (~s-mutin@mail.ska-irk.ru) |
| 16:10.17 | CRCinAU | so here's another strange one. |
| 16:10.39 | CRCinAU | how can I specify which sound file packages to install without using make menuselect? |
| 16:11.19 | file | https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options#UsingMenuselecttoSelectAsteriskOptions-ControllingMenuselect |
| 16:11.57 | CRCinAU | oooooo |
| 16:15.21 | *** join/#asterisk k3asd` (~k3asd@unaffiliated/k3asd/x-0305612) |
| 16:16.18 | CRCinAU | I thought I was going to have to break out some *nasty* sed stuff :P |
| 16:16.22 | CRCinAU | that's much nicer. |
| 16:20.26 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 16:25.27 | igcewieling | CRCinAU: I use this: menuselect/menuselect --enable=format_mp3 --enable=cdr_mysql --disable=CORE-SOUNDS-EN-GSM --disable=MOH-OPSOUND-WAV |
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| 17:21.31 | CRCinAU | right - so another fun one..... |
| 17:21.39 | CRCinAU | so I'm trying the en_GB sounds..... |
| 17:21.48 | CRCinAU | they exist in /var/lib/asterisk/sounds/en_GB/ |
| 17:21.52 | CRCinAU | but nothing seems to find them |
| 17:22.08 | CRCinAU | what's the 'proper' way to fix this (ie not moving en_GB -> en) |
| 17:24.56 | CRCinAU | ah. |
| 17:25.15 | CRCinAU | I just did a #yolo and added: language = en_GB on just about everything lol |
| 17:25.56 | CRCinAU | now its 3:25am and I need sleep. |
| 17:26.24 | igcewieling | look in asterisk.conf.sample for "Use the new sound prefix path syntax." option |
| 17:26.46 | aiksa[LV] | no, true #yolo is updating kernel with randconfig |
| 17:26.54 | CRCinAU | heh ;) |
| 17:27.05 | aiksa[LV] | and expecting it to boot up |
| 17:27.39 | CRCinAU | like rebooting a headless / no KVM / unattended DC server |
| 17:27.45 | CRCinAU | it should come back up. |
| 17:28.12 | CRCinAU | anyhow - sleep.... its stupidly lat... err early.... |
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| 22:01.22 | k3asd` | hi there, I'm trying to configure encrypt calls via asterisk with srtp. I'm following this tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 22:01.46 | k3asd` | in the asterisk cli I receive: [Apr 11 18:01:16] NOTICE[10628]: res_pjsip_transport_management.c:136 idle_sched_cb: Shutting down transport 'TLS to 78.15.226.36:40652' since no request was received in 32 seconds |
| 22:01.56 | k3asd` | in my blink softphone: registration failed |
| 22:02.10 | k3asd` | how can I fix this issue? thanks in advance |
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