IRC log for #asterisk on 20160509

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04:37.16IG99Hi folks. What are some of the asterisk monitoring software out there? I want to plonk it on a central server and have it monitor 2 of our PBXs. Any good ones? Thanks
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05:15.31FarhaadNcan u please help me about this issue > https://community.asterisk.org/t/asterisk-13-with-sipml-without-audio/66645
05:40.40cmendes0101FarhaadN: Is this working with a normal SIP client?
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05:43.17cmendes0101FarhaadN: What are your settings in sipml5? seems like it has the 192 IP set in there
05:43.50z30Hey guys, i have problem with asterisk ooh323 and need help plz, i have one way audio
05:44.45z30two asterisk connect to each other via ethernet and Call from A to B is OK but when i call from B to A i have no audio
05:45.04z30same Setting and configuration for both
05:45.40z30what can i do?
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05:56.04FarhaadNcmendes0101: thx for reply , is worked with normal sip client
05:57.13cmendes0101probably something on the sipml5 config
05:57.13FarhaadNcmendes0101: i solved issue by disabling all my network interface,i don't know why ICE condicate with other interfaces
05:57.19cmendes0101oh
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05:57.30cmendes0101ok cool
05:57.47FarhaadNcmendes0101: but audio issue still exist
05:58.24FarhaadNICE candicate is now correct ip but there is no auido
05:59.31cmendes0101yah, whats the config for sipml5?
05:59.40FarhaadNany config
05:59.50cmendes0101theres only 1
06:00.07FarhaadNi used demo,but no audio
06:00.17cmendes0101ok, whats the config you entered in the demo lol
06:00.29cmendes0101the boxes with settings on the site
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06:00.38FarhaadNi change nat,stun, and any of field
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06:02.32FarhaadNfill ICE server with : []
06:02.45FarhaadNwebsocket  server URL : wss://172.17.106.8:8089/ws
06:05.36FarhaadNmy client ip : 172.17.106.1112
06:05.37FarhaadNmy server ip : 172.17.106.8
06:05.37FarhaadNrtp log in asterisk :
06:05.37FarhaadNSent RTP packet to      172.17.106.112:50165 (via ICE) (type 00, seq 018040, ts 060480, len 000160)
06:05.37FarhaadNSent RTP packet to      172.17.106.112:50165 (via ICE) (type 00, seq 018041, ts 060640, len 000160)
06:06.02FarhaadNthere is no recive RTP packet from ICE
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09:33.36*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.2 (2016/04/20), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
10:01.43post-factumguys, need help with tls, please. i've got an account on SIPNET, and now trying to use it for TLS+SRTP
10:01.48post-factumhere is sip.conf section: http://termbin.com/ngjs
10:01.53post-factumbut on sip reload I get the following error: http://termbin.com/eyem
10:01.57post-factumwhy that happens?
10:04.31Zogotpost-factum: im not entirely sure, but is this related to the register => line you have
10:04.41Zogotyou could also relate it to the sip peer directly
10:05.17Zogotpost-factum: change your register line to this
10:05.42post-factumZogot: but i do not have any register => line, as sipnet docs clearly says it is not necessary to have that
10:05.51Zogoto
10:05.55Zogothmm
10:06.21post-factumZogot: i assume peer definition also enables registration
10:07.15Zogotpost-factum: ive had to define register lines before, i often have them reference the peer directly
10:07.31Zogotlike register => transport://username@sippeername/number
10:07.44Zogotyour tls options are set under sip.conf [general]?
10:08.42post-factumZogot: let me show you the whole sip.conf...
10:10.06Zogotpost-factum: ensure everything you paste has retracted values
10:10.44post-factumZogot: for sure :) http://termbin.com/fsf9
10:11.01post-factumZogot: with this config everything is unencrypted and works ok
10:11.25post-factumZogot: if i change insecure-trunk to secure-trunk for peer definition, it doesn't work
10:12.37post-factumZogot: (i mean template)
10:13.12Zogotpost-factum: have you tested that all your certs and stuff are right with trying to register a normal peer?
10:13.24Zogotfrom a softphone or so
10:13.43post-factumZogot: yep, my csipsimple from android registers and works well
10:13.56post-factumZogot: it shows SIPS and SRTP enabled while making the call
10:15.38post-factumZogot: here is how i define user if that matters: http://termbin.com/yqgd
10:16.47post-factumZogot: also, certs are not self-signed, those are real certs for server domain
10:17.14Zogotpost-factum: hmm you could try to simplify it, getting it closer to reference material and tweak from there
10:17.31Zogotexamples are things like insecure=port,invite instead of just invite, media_encription values?
10:17.39Zogotremoval of templates and so on
10:17.50Zogotnot the best at this yet, but thats how id go about solving it
10:18.12post-factumZogot: removing templates didn't help :). have tried it already
10:18.19Zogotcan't say ive seen your error before, but ive not tried tls trunks yet
10:18.38Zogotthe only reading ive done, is related to twilio, and they have this document, with examples at the bottom: https://www.twilio.com/resources/images/docs/Asterisk-Twilio-Secure.pdf
10:18.45Zogotmaybe there it could help you
10:19.07post-factumZogot: shouldn't media_encryption be for pjsip?
10:19.22Zogottheres some values there, that you dont have, different providers obviously, but perhaps might be something to read into
10:19.29Zogotumm
10:20.21Zogotpost-factum: might be right, ctrl+f encryption only shows what you expect: http://svn.asterisk.org/svn/asterisk/branches/13/configs/samples/sip.conf.sample
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10:22.07Zogotpost-factum: a good note to read in the sip.conf.sample
10:22.14Zogotpost-factum: ctrl+f for tlsbindaddr
10:23.13Zogotpost-factum: maybe you should try adding a register => tls://
10:23.23Zogotpost-factum: thats what i would of read your original error about
10:23.55Zogotanyway, im about to go eat, hope any of this helps :p
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10:26.24post-factumZogot: how dedicated register line differs from peer declaration?
10:26.31post-factumZogot: what is the difference at all?
10:27.05Zogotpost-factum: ive always had both
10:27.12Zogota peer definition and a register line
10:27.24Zogotpost-factum: the register line goes under [general]
10:27.41Zogotregister => tls://username@peer/number
10:27.51post-factumZogot: anyway, adding register line doesn't change anything. error still appears
10:28.48Zogotpost-factum: maybe if you turn sip debug on, the packets will say something
10:29.24FarhaadNcan u help me about this issue > https://community.asterisk.org/t/asterisk-13-with-sipml-without-audio/66645
10:32.46post-factumZogot: nothing there :(
10:35.58Zogotpost-factum: you have tlsdontverifyserver=no?
10:36.16Zogotpost-factum: maybe its a port thing, ive not messed with that much
10:37.16Zogothttp://wiki.vpsget.com/index.php/Asterisk_11_with_TLS_and_SRTP_on_Centos_6 that uses sipnet as the example
10:39.20Zogotanyway, off to eat
10:45.42post-factumZogot:  tlsdontverifyserver=no ← tried changing that with no luck as well
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11:27.07post-factumZogot: here?
11:28.22Zogotpost-factum: hey, im back
11:28.52post-factumZogot: i've narroed down the issue a little bit. will you take a look?
11:28.57post-factum*narrowed
11:29.36post-factumZogot: lets say we have this peer definition in sip.conf: http://termbin.com/klae (no register line, just this)
11:29.37Zogotpost-factum: sure man
11:30.02post-factumZogot: then new entry appears in sip show registry output: http://termbin.com/royv
11:30.14post-factumand it is shown as 5060, despite we have set transport to tls!
11:30.19post-factumwhat the f?
11:30.38Zogotpost-factum: i must say, ive never seen a registry line without a register line somewhere
11:30.39post-factumeven changing port to 5061 forcely does not change sip show registry output
11:31.07Zogotits not possible that a register line exists somewhere?
11:31.12post-factumnope
11:31.58Zogotpost-factum: callback-extension?
11:32.13Zogotah nothing
11:32.18Zogotjust looking at your sip peer
11:32.21Zogotwhat its like to mine
11:33.17Zogotah it is it
11:33.19Zogotpost-factum:
11:33.22Zogotits the callback extension
11:33.26Zogotget rid of it, add a register line
11:33.39post-factumhmmmmm
11:33.55Zogotadd the register line as tls:// blabla
11:34.07post-factumok, give me a moment
11:34.54post-factumwhat extension should i set for register line?
11:34.58post-factumafter last /
11:35.42Zogotpost-factum: how are you structuring your register, as tls://username@peername or tls://username:password@host ?
11:36.20post-factumtls://username:password@host
11:36.31Zogotok, so it'll use the context defined in your [general] in sip.conf
11:36.39Zogot[general] context=example
11:36.44Zogotyou then want an extension that exists in example
11:37.11post-factumshould i set password twice, in both register and peer declaration?
11:37.31Zogotyes if you do it your way
11:37.37Zogotits why i like doing it the @peer way
11:37.52Zogotanother benefit you have with @peer way is it uses the context defined in the peer
11:38.02Zogotcan help with multi tennant use cases for example
11:38.10post-factumbut now i have both register and peer
11:38.17Zogotyep thats correct
11:38.17post-factumwhich context will it use?
11:38.27Zogotif you are using @host [general] in sip
11:38.35Zogotif you are using @peer then the context in the peer
11:38.48Zogot@host context in [general]
11:38.50Zogotsorry
11:39.34post-factummeh, still did not get that :(
11:39.45Zogotok, so i'll give you an example if you give me 2 minutes?
11:40.11post-factumyep. sure
11:41.04Zogotpost-factum: il show you how i do it, with an explanation of why
11:41.11Zogotbut i'll give the common example also
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11:45.08post-factumZogot: oh, it seems if i do not put extension in registry line, it uses context set in peer definition
11:45.13post-factumZogot: and that is fine for me
11:46.27Zogotpost-factum: ah :)
11:46.52Zogotpost-factum: https://gist.github.com/zogot/8d33319490df6bc408bd134591e529c4
11:46.54Zogothere ya go
11:48.19Zogotpost-factum: chances are you have 's' in the context in your peer
11:48.29post-factumZogot: so if call comes from registered peer, it will go to the context defined in peer definition
11:48.32Zogotwhich asterisk falls back to as the extension if it does'nt know what to do
11:48.36Zogotpost-factum: yep
11:48.39post-factumZogot: if it is unregistered call, it will go to default
11:48.42Zogotif you do register @peer
11:48.52Zogot[general] context
11:49.01Zogotif unregistered (only possible from allowguest=yes)
11:49.07Zogotelse asterisk stops it anyway
11:49.34Zogotpost-factum: this also cleans up a lot for situations where trunks are load balanced
11:49.39Zogotsince they can come from multiple ips
11:49.54Zogotusing @peer will check against the host
11:50.01Zogotand you wont have issues with multiple registers
11:50.24Zogot(and thus wont need allowguest=yes) its what we had in our of our versions of software until i read about @peer
11:50.48post-factumZogot: so register <=> peer matching is done by user@host
11:50.56Zogotuser@peer
11:51.01Zogotin the register line
11:51.11Zogotuser@trunk_peer_name_in_square_brackets
11:52.44Zogotpost-factum: did you stop getting the error at least :p
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11:53.19post-factumZogot: yep, everything seems to work now, but you confused me even more :)
11:53.36post-factumZogot: let me check another example, and you'll tell me if i think correctly
11:53.42Zogotok
11:55.25post-factumZogot: in this case http://termbin.com/k5ox
11:55.53post-factumZogot: all incoming calls from authenticated peer will go to from-example context to s extension?
11:56.09Zogotnope, it'll go to the context defined in [general]
11:56.11Zogotin sip.conf
11:56.38post-factumok, how one would make it go to from-example context?
11:56.57Zogotchange register to 12345@example
11:58.23post-factumor change [example] to [example.com]?
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11:58.42Zogotpost-factum: but iirc, its recommended to not do that
11:58.47Zogoti think it causes other issues
11:58.49post-factumwhy?
11:59.23Zogotbecause i would imagine it would be attempting to determine if its a peer or a remote url by that format
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11:59.42Zogoti think the allowed characters for contexts are A-Za-z_-0-9
11:59.45Zogotits pretty simple
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11:59.59Zogotfor what they allow
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12:00.39post-factumwell, i need to test that now
12:00.59post-factumwhat i see now is that it goes to right context despite having [example]
12:01.18ZogotContexts are defined by placing the name of the context inside square brackets ([]). The name can be made up of the letters A through Z (upper- and lowercase), the numbers 0 through 9, and the hyphen and underscor
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12:01.57Zogotdespite having [example]? you mean in the register?
12:02.20post-factumagain, this case: http://termbin.com/k5ox
12:02.34post-factumall incomings via this peer goes to from-example context
12:02.40post-factum*go
12:03.19post-factumdespite having context=invalid in [general]
12:09.53Zogotpost-factum: umm that is odd, you core reload 'ed after changes right?
12:10.03post-factumyep
12:17.39post-factumfile: any ideas on tls registration without register line?
12:17.51filenope, I've never done it in chan_sip
12:18.08post-factumfile: should new installation consider using pjsip instead?
12:18.36filePJSIP is core supported in 13 and above, chan_sip is in extended support so we (Digium) are putting our focus on PJSIP
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12:20.02post-factumfile: and chan_sip is planned to be dropped somewhen in the future?
12:20.11fileit is unlikely it will ever be dropped
12:20.21post-factumokay
12:22.46filebut from a resource perspective we (Digium) won't be putting effort towards it in 13 and above, but if community members submit changes we'll review them and such
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12:26.29post-factumfile: i see, thx
12:26.44post-factumZogot: so i tested it again, and yes, it goes to right context
12:26.54post-factumZogot: so i guess matching is done by user-host pair
12:27.04post-factumfile: am i correct with that?
12:27.29post-factumfile: http://termbin.com/k5ox here what is the relation between register line and peer definition?
12:27.35filechan_sip does user matching of the From first, followed by IP based+port matching if you enable it, or just IP if you enable it
12:28.38post-factumdoes that depend on "insecure" option?
12:28.55filethat controls IP/IP+port based matching
12:29.14post-factumnow it is much more clear to me
12:29.21post-factumthanks, file
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14:40.06*** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.0 (2016/05/09), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
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14:50.07catphishi'd just like to take this moment to vent my frustration at an upstream SIP provider who don't support "180 Ringing" and only send in-band ringing :(
14:51.01filethat can be legit, depending on the end switch
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14:52.29catphishit's quite unhelpful though :(
14:53.28catphishit's ok for a 1:1 call, i can just pass through the in-band ringing, but for a multi-destination call, there's not really any way to know when to start playing ringing to the caller :(
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17:15.03ChkDigitDoes anyone know how much current a Digium A4 card draws over the AMP connector?  Or more specifically, if it is within the limits of a 22ga floppy drive connector?
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18:02.15newtonrChkDigit, Unsure at this moment but I can get you an answer. I'll E-mail an engineer and see if I can find out. PM me your email and I'll get you a response eventually.
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19:20.55MajesticFudgieAnyone in here familiar with Polycom phones?
19:21.32[TK]D-FenderLots of people
19:23.00MajesticFudgieAh, didnt know how popular they were
19:24.12MajesticFudgieI have an issue (aside from the fact its stuck in a reboot loop right now -_-) where it can recieve incoming calls but picking up the handset makes it flash "URL Calling Disabled" on the screen. I have a Polycom Soundpoint 331 if that makes any difference.
19:24.22MajesticFudgieIs this common issue with VOIP Phones?
19:24.41[TK]D-Fenderno.
19:24.52[TK]D-Fendereven having that as a concept isn't unioversal
19:25.03[TK]D-FenderNothing should make it bot look like that unless something seriously flakey is going on.
19:25.12[TK]D-FenderHow did you congiure it?
19:26.06MajesticFudgieOriginally on the Phone its self. and nothing worked at all.
19:26.27MajesticFudgieThen I followed a guide by Polycom on setting up a boot server which is started using
19:26.41MajesticFudgie*I started using
19:27.02MajesticFudgieSo now it loads the config off a HTTP Server, along with sip.id
19:28.39*** join/#asterisk vader- (~Adium@50.232.174.194)
19:29.32MajesticFudgieOh sip.ld
19:32.50[TK]D-FenderYou should disable URL dialing in your provisioning
19:33.01MajesticFudgieIdk how to
19:33.42MajesticFudgieand I hope I have the right firmware. I read that the rebooting could be a firmware issue
19:34.31[TK]D-Fenderthe firmware itself wouldn't be, only the config you generated for it
19:34.35[TK]D-Fenderso fix that first.
19:34.53[TK]D-FenderConfigs are easily searchable for that option
19:34.57[TK]D-Fendergo kill it off
19:35.03[TK]D-FenderAnd verify what you're running on it
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19:39.36MajesticFudgieWell, It says Revision 4.2.1.0275, 16-Nov-2009
19:42.17MajesticFudgieIs the Provisioning server and Boot Server the same thing just different terminology?
19:46.06[TK]D-FenderThe SIP version
19:46.09[TK]D-Fendernot the bootrom
19:46.34[TK]D-FenderMajesticFudgie> Is the Provisioning server and Boot Server the same thing just different terminology? <- yes, this i referring to the same thing
19:46.46MajesticFudgieAh
19:47.00MajesticFudgieand thats all I can find from the main menu on it before it boots
19:47.04MajesticFudgieas soon as it boots it restarts
19:47.19MajesticFudgieIt flashes the 'home screen' for a split second before "Reboot Initiated"
19:50.28[TK]D-FenderYou mentioned getting calls on it, and now just looping
19:50.30[TK]D-FenderWhich is it?
19:51.32MajesticFudgieboth tbh, It started to be able to only get calls and now its looping
19:52.12[TK]D-Fenderif it's looping then you have a serious config error
19:52.15[TK]D-Fendertypically
19:54.20MajesticFudgiehmkay
19:54.24MajesticFudgieAny way to resolve that?
19:54.35[TK]D-FenderFix the error
19:54.35MajesticFudgieAs the phones menu before booting is limited
19:54.43[TK]D-Fenderand have the phone pick it up
19:54.59MajesticFudgiemkay
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20:06.08MajesticFudgieI see Rev: 3.2.1 when it boots actually
20:06.16MajesticFudgieidk if thats the stuff you're looking for
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20:07.23MajesticFudgieRight I've got it to log to the provisioning server, I see its saying the sip.ld file isn't compatible
20:07.37[TK]D-Fender3.2.1 is pretty old
20:08.11[TK]D-Fenderhttp://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
20:08.34MajesticFudgieApplication, load: Type=SIP, Version=3.2.1.0156 03-Dec-09 21:33
20:08.38MajesticFudgieIs in the logs
20:08.46MajesticFudgiesec
20:09.21MajesticFudgieWhich do you recommend? Combined or Split?
20:09.37MajesticFudgieAs I were trying to use the latest it shows, the combined one and it says its not compatible
20:11.16post-factumhmm, calls going from udp trunk to tls phone end up with "busy"
20:11.59post-factumthat is now how i imagine working tls for voip :(
20:11.59[TK]D-FenderCombined
20:12.03post-factum*not
20:13.04[TK]D-Fenderpost-factum, the peieces of what you wrote there do not show what actually happened
20:13.26post-factumah ok. here it is:
20:13.38post-factumi have ast 13 server with tls enabled for sip phones
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20:13.51[TK]D-Fenderpost-factum, No more than "I had a car.  Then I crashed. This is not how I imagined cars to work. "
20:14.30post-factumalso i have sip trunk to sip provider handling incoming calls and routing them to one of tls phones
20:14.54[TK]D-Fender"route" would not be appropriate.  There is just "dial" (or queue, etc)
20:15.02post-factumbut after incoming call happens, it goes to busy circuit with "SIP/2.0 480 Temporarily Unavailable" and "Warning: 381 localhost "SIPS Required""
20:15.05[TK]D-FenderSo lets look at the failure itself
20:15.24[TK]D-FenderALL of it.
20:15.30[TK]D-Fendernot just a 1-line snippit
20:16.18post-factumshould i post sip debug output for tls peer?
20:18.05[TK]D-Fenderfor BOTH ends
20:18.15[TK]D-FenderIf you're not looking at the whole call then you are running blind
20:20.09MajesticFudgieThe phone says sip.ld is not compatible with phone .-.
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20:22.45post-factum[TK]D-Fender: https://gist.github.com/d15d427a1272222ab452513010947eff
20:23.45post-factumhmm
20:23.47post-factumno
20:23.59post-factumthat is not the whole log, i do not see busy message there
20:27.43post-factumok, this one https://gist.github.com/500894a3d1e1dae511c75223c8198ce1
20:28.15post-factumline 291 contains the errors i posted above
20:31.53[TK]D-FenderNo need for core debug in there and we are missing basic verbose.  You hsould also show your peer for the TLS one masking only the secret.
20:32.00[TK]D-FenderAnd I'll have to pick this up when I get home.
20:32.07[TK]D-Fenderheads out from the office...
20:35.50Kevin`how can I tell asterisk to autenticate during invite?
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20:51.02post-factum[TK]D-Fender: sorry, no idea how to gather logs properly now...
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21:14.41[TK]D-Fenderpost-factum, :
21:14.45[TK]D-Fendercore set debug 0
21:14.51[TK]D-Fendercore set verbose 10
21:14.59[TK]D-Fendersip set debug on
21:15.15[TK]D-Fenderor if using pjsip : pjsip set logger on
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21:30.08post-factum[TK]D-Fender: ok, give me a moment, i've prepared test setup from scratch
21:30.30[TK]D-Fender"from scratch"?
21:31.18post-factumyep, to throw away all the garbage from config to narrow issue field
21:34.29MaliutaLapwell I finally found a carrier to do .ca DID's at a reasonable cost ... pity it's about 5 years too late
21:34.37Kevin`any help would be appreciated, this issue is hard to find information on since everyone solves it by using registration and insecure=invite
21:34.41post-factumso
21:34.46Kevin`as such it's not searchable
21:34.49post-factum[TK]D-Fender: here is my sip.conf: http://termbin.com/eok6
21:35.06post-factum[TK]D-Fender: here is my log https://gist.github.com/b964a15c77eda0c78d161715dcdeec68
21:35.57post-factum[TK]D-Fender: calling from 302 (transport udp) to 301 (tls) is impossible. calling from 301 to 302 goes ok
21:36.19post-factum[TK]D-Fender: and I wonder why
21:37.01post-factum[TK]D-Fender: line 226 is with 480 error
21:42.33post-factum[TK]D-Fender: i see similar sympthoms here: https://issues.asterisk.org/jira/browse/ASTERISK-24646 but i do not use pjsip
21:52.39[TK]D-Fender<[TK]D-Fender> core set verbose 10 <---------------------
21:53.52post-factumi did that!
21:54.09[TK]D-FenderThen I'd have seen the DIAL executing from CLI
21:54.26[TK]D-Fenderyou ***ARE*** doing this capture from live CLI, aren't you?
21:54.38[TK]D-FenderLogs = garbage
21:54.41[TK]D-FenderCLI = everything
21:55.19post-factumok, one moment
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21:58.51post-factum[TK]D-Fender: https://gist.github.com/8526416517a8d42e2a31462cd98253dd
22:00.44[TK]D-Fender[May 10 00:56:05] ERROR[882]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:1407609B:SSL routines:SS
22:00.44[TK]D-FenderL23_GET_CLIENT_HELLO:https proxy request
22:00.44[TK]D-Fender[May 10 00:56:05] WARNING[882]: tcptls.c:684 handle_tcptls_connection: FILE * open failed!
22:00.54[TK]D-FenderCertainly doesn't look good...
22:01.52post-factumrepeated 3 more times, and do not see that error
22:02.59post-factum[TK]D-Fender: https://gist.github.com/edde24f5d30400133a0ee3856c5d16c5
22:03.08[TK]D-Fender"module reload chan_sip.so"
22:03.14[TK]D-Fenderand "sip show settings"
22:03.16[TK]D-FenderPB it all
22:03.46[TK]D-FenderAnd show the client config
22:04.41post-factum[TK]D-Fender: https://gist.github.com/e040dd08cbd792dafc85e057c1600b8b
22:06.02post-factum[TK]D-Fender: tls client is csipsimple, unencrypted client is linphone…
22:12.34[TK]D-Fender<[TK]D-Fender> And show the client config
22:13.11post-factum[TK]D-Fender: csipsimple screenshots: http://goo.gl/qd4rCy http://goo.gl/BjkqQj http://goo.gl/V6IECW http://goo.gl/BQLDDd http://goo.gl/6wy5yS http://goo.gl/deoEHN http://goo.gl/6OzqaB
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22:15.43[TK]D-FenderI'm wondering on SRTP requirement there
22:17.17post-factummandatory?
22:18.06post-factumchanged to default and optional — no changes
22:18.46post-factum«disabled» — the same
22:19.38[TK]D-FenderI'm a little short on ideas here myself not having really played with TLS/SRTP before.
22:19.57[TK]D-Fenderperhaps someone else will be able to run beter with what you've now provided.
22:20.00post-factum:(
22:20.24[TK]D-Fenderbe patient and someone more experienced should be able to carry this forward with you
22:20.57[TK]D-Fenderfile, mjordan, newtonr : any of you able to chime in with the TLS issue he's provided good backup for?
22:21.44fileI don't speak chan_sip in that area.
22:23.12filelooks like whatever proxy is in the mix is probably unhappy with the From that has no sips
22:23.32fileor the Contact too
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22:31.03fileproxy = remote side, dunno why I said proxy.. although I have been in that land today
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22:39.19V5_12R\XSo can anyone tell me where the sounds path is set?
22:40.55V5_12R\Xasterisk is failing to start looking for the path /var/lib/asterisk/sounds/en/sounds... which I cant find set anywhere...
22:41.56V5_12R\Xnevermind, it's that the dir existed
22:42.34V5_12R\XI swear one day I'm going to just document every way that I break this thing
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22:46.04[TK]D-FenderAfter a certain point it may be shorter to write about the way's you haven't..
22:47.04V5_12R\Xpost-factum I am guessing that this is because the call is being forwarded. Certs need to be set up for all parties, so I am guessing (although I dont know chan_sip that well with SSL\TLS) that this going to be a cert issue with one of the legs of the call.
22:48.20V5_12R\X[TK]D-Fender however less useful to those needing assistance
22:48.58[TK]D-FenderShouldn't need to
22:49.04V5_12R\Xas in it would have been helpful to know that ast chokes on sub-directories of /var/lib/asterisk/sounds/en/
22:49.05[TK]D-Fender* is a B2BUA
22:49.23V5_12R\Xso dont do that
22:51.17V5_12R\Xunless direct media is involved...
22:51.29V5_12R\XI'd like to see if it works with just one hop
22:55.54*** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net)
22:56.05MiccHow do I make a queue not ring members who are already ringing?
22:57.36[TK]D-FenderTell it not to in the queue definition
22:59.09MiccWould this be the autofill?
22:59.16[TK]D-Fenderno.
22:59.26[TK]D-FenderThat is not at what the sample config says it's for...
23:12.02MiccSo what is the proper setting?
23:12.41Miccringinuse is only for phones "in use" not ringing.
23:12.47Miccso setting it to no doesn't help
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23:18.59[TK]D-Fenderringing = inuse
23:19.16[TK]D-FenderShow us your now proper config, the queue memeber dump and the call
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