IRC log for #asterisk on 20180921

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01:58.18*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Standard: 15.6.1 (2018/09/20); 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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11:11.14Jimmy06Hey guys wondering if you might be able to point me in the right direction to add my own AMI call
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11:29.15Jimmy06I simply need a custom version of message
11:32.07fileit depends on the version, and there isn't a central place for it
11:34.38Jimmy0613.20
11:34.58Jimmy06I need to add say Event: and change the Content-Type
11:35.23filethat doesn't provide enough context, are you wanting to change an existing event?
11:35.48Jimmy06I'd rather add my own event based on the messagesend event
11:36.13filehttps://github.com/asterisk/asterisk/blob/master/main/channel.c#L6604
11:36.21filethat is an example of the easiest way to send an AMI event
11:36.38Jimmy06http://wiki.snom.com/FAQ/How_to_change_settings_using_SIP_message
11:36.57Jimmy06Im trying to do this basically
11:37.48fileso you want to add an AMI action?
11:37.57Jimmy06yes
11:38.15filethere isn't a guide or anything, the easiest option would be to find an existing AMI action and look at how it works
11:40.11Jimmy06where is the ami messagesend action is it here https://github.com/asterisk/asterisk/blob/fcd8e9d6ef6cad66ba3bfe58540aa1cdb372fc2c/main/message.c
11:41.15fileyes.
11:42.15Jimmy06what would be the easiest way to send that event. we cant use sipsak as the phones are not local
11:42.50fileit's not an event, it's an action
11:43.05fileunless you are referring to the SIP message earlier?
11:43.38fileif it's strictly a MESSAGE, then support was added recently to set such things - I don't know if it landed in 13
11:43.41Jimmy06yes the sip message
11:43.46Jimmy06not the AMI action
11:44.04fileI have to go look, lemme see
11:44.25fileadded as of 13.22.0
11:44.52filehttps://github.com/asterisk/asterisk/blob/13/CHANGES#L42
11:45.37Jimmy06I cant update the version of asterisk that's a shame
11:45.47Jimmy06wonder if I can backport it in
11:46.05fileI don't know if we allowed application/xml as a content type
11:46.28fileanyway, the change could be a basis for what you'd want to do
11:47.07Jimmy06Yes. Thank you
11:49.23Jimmy06thats the standard SendText action right ?
11:49.53filedialplan application or AMI action, I didn't look deeper
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11:51.33Jimmy06its Dialplan for context type
12:00.23Jimmy06file: would it be easier to just create a technology ?
12:00.53fileno? a technology is just the interface between the core and a channel driver
12:08.30Jimmy06I dont see what part of messagesend sets the content type though
12:08.48fileare you looking at your old version?
12:09.27Jimmy06messagesend not sendtext
12:09.58Jimmy06https://github.com/asterisk/asterisk/blob/fcd8e9d6ef6cad66ba3bfe58540aa1cdb372fc2c/main/message.c
12:10.04Jimmy06where in this is the content type set
12:11.08filethat interface may not have gained the ability to set it
12:11.17filein which case it would be defined in the technology implementation
12:12.21Jimmy06hmm ok
12:12.47SamotUhm....
12:12.52Jimmy06so there is no easy way to send a custom sip message to a phone?
12:12.53SamotWhat are you exactly trying to do?
12:13.04Jimmy06http://wiki.snom.com/FAQ/How_to_change_settings_using_SIP_message
12:13.05filewith a custom content-type? seemingly not using that, no
12:13.07Jimmy06this basically
12:13.08fileyou'd have to do code modifications
12:13.14SamotYou want to send a SIP MESSAGE to an IP Phone?
12:13.27Jimmy06yes via asterisk
12:13.35fileSamot: with a custom content-type and XML payload.
12:13.41Jimmy06^^
12:15.02SamotSounds like kamailio.org is needed
12:16.08SamotBecause this isn't going to be something that Asterisk does.
12:16.29SamotI guess it could if you really wanted to dick around with the source and make all sorts of modifications.
12:17.05SamotOr you can just get a SIP Proxy, like Kamailio, that is designed to tear apart, rebuild, create, etc SIP packets/request/replies.
12:17.29SamotYou could easily create your MESSAGE with the xml content-type and payload.
12:17.54Jimmy06the phones are offsite so firewalled
12:18.12Jimmy06firewalled/NATed
12:18.45SamotOK
12:19.00Jimmy06so a sip proxy/kamailo is probably the answer
12:19.04SamotLuckily a lot of the proxies are just for that fact.
12:19.40Jimmy06Right OK any suggestions on a decent SIP Proxy ?
12:20.29Jimmy06sipproxd ?
12:20.32SamotUhm
12:20.40Samot8:15:02 AM <Samot> Sounds like kamailio.org is needed
12:20.47Samot8:17:04 AM <Samot> Or you can just get a SIP Proxy, like Kamailio, that is designed to tear apart, rebuild, create, etc SIP packets/request/replies.
12:21.01Jimmy06right let me look at that then
12:21.22Jimmy06Sorry distracted :-) THank you
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16:09.50kombiyour highnesses, how do I prevent a ringgroup from going to each extension's voicemail?
16:20.55igcewielingkombi: as far as I know calls to ring groups don't go to voicemail unless configured.
16:21.27igcewielingI assume since you are on #asterisk and not #FreePBX you are not using a GUI.
16:22.16fileAsterisk doesn't also have a concept of ring groups...
16:22.43igcewielingfile: good point.
16:23.01igcewielingI suspect he is using a GUI and not telling us.
16:23.17fileI suspect... bunnies.
16:25.55kombitouché, I did as a vi person for the first time set up a box with again GUI, so my comrades in the office comprehend the configuration. I didn't realize ringgroup were not an asterisk feature, sry!
16:27.32SamotA Ring Group is just a method/logic for calling a group of endpoints in a specific manner.
16:28.03SamotThere isn't a RingGroup() app or function like Queue() or Park()
16:29.00SamotYou have to build it how you want it to work. There are some standard setups that most use for this but when it gets down to fancy features like call confirmation, etc, etc that has to be coded...
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16:29.23SamotAlso, don't have the ring group call the extensions for so long they trigger the voicemail.
16:29.35SamotThat's basically the answer to your question.
16:30.18kombiI see... so that is actually implemented in that GUI... I do regret using that. Question answered profoundly, thanks everyone!
16:30.19SamotDoesn't matter what PBX you are using, if the Ring Group is set to ring the members for 25 seconds and a member has their VM set for 10 secs...
16:30.27SamotThat member is answering pretty much _every_ call.
16:30.58SamotWhat GUI is this?
16:31.20kombiI hate to admit, FreePBX..
16:31.26SamotUhm.
16:31.35SamotSince FreePBX owns Digium now...
16:31.52SamotAnd the Ring Groups in FreePBX have all sorts of options to avoid this kind of stuff...
16:32.00SamotYou have your stuff configured porrly.
16:32.04SamotYou have your stuff configured poorly.
16:32.31SamotWell Sangoma, not FreePBX.
16:32.51kombiI heard, but will that improve things I wonder. I considered draping that installation a number of times
16:33.05kombi*scraping
16:33.05SamotAlthough the frog does insist that @file call him MR. Tango.
16:33.18filewhat what what
16:33.22SamotHAHAHA
16:33.59filetechnically it's Sangoma owning Digium, not FreePBX as FreePBX refers to the project
16:34.14SamotI did correct myself.
16:34.14filewhich I see you corrected
16:34.18Samot:)
16:34.50Samotkombi: I would review your Ring Group settings. They are probably misconfigured.
16:35.16kombiI I understand it right FreePBX got bought by sangoma and proprietary stuff for their phones was added, now they also bought asterisk and digium. Will they phase out digium phones? I just bought a load of D80s..
16:35.26SamotBut the Ring Group "Ring Time" needs to be less than members "Ring Timer"
16:35.39SamotNo.
16:36.22Samothttps://wiki.freepbx.org/display/FOP/Sangoma+and+Digium+Join+Together+FAQ
16:36.24kombiSamot: That is good news, also I will look at the number of rings
16:36.54SamotThe FAQ covers most those types of questions.
16:36.55fileif you have questions in that regard then contacting your account manager would be appropriate per the FAQ
16:48.35SamotAlthough, FreePBX did make me sad last night.
16:50.05SamotI wanted to start using PJSIP for systems/endpoints that aren't required to run Chan_SIP for outside program needs but FreePBX always assumes "registration" for the aor section on the endpoints.
16:50.36SamotThere's no way to set static contact URI's for the endpoint.
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16:50.57fileah
16:51.37SamotWhich sucks because, Kamailio. The systems all think the endpoints are Kamailio.
16:53.21SamotSo I'm going to need to make Kamailio send REGISTER requests with the auth headers and expire attributes set to make Asterisk bypass the 401 challenge and register the contact.
16:54.28SamotYay....
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17:04.04igcewielingSamot: are you saying pjsip doesn't support static IPs for endpoints when not using registration?
17:05.10igcewielingI have something like 290 peers and 4 or 5 of them register, the rest are static IPs and setup without registration.
17:05.48filePJSIP does.
17:14.44SamotNo, that's not what I'm saying.
17:15.03SamotI'm saying that FreePBX does not support assigning static contact URI's to extensions.
17:15.20SamotIt always assumes "registration" for the contacts on the endpoint.
17:15.51SamotIf I setup a PJSIP endpoint on my own, it works great because I can set static contacts.
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18:03.06shanth_is there a way to automatically log everything in the asterisk console to a file?
18:04.16rmudgettshanth_: That's what the configuration in logger.conf does.
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18:05.52shanth_nice thanks rmudgett
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