IRC log for #asterisk on 20170709

00:19.30*** join/#asterisk infobot (~infobot@rikers.org)
00:19.30*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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
00:44.30UncleKiwiit would be cool if a whatsapp channel driver could be made - does anyone know of a way i could forward a call from asterisk to Whatsapp ?
00:44.55UncleKiwisimilar to the old 'skype channel driver'
00:58.33SamotSkype wasnt encrypted
00:59.04UncleKiwiah ok
00:59.20SamotPlus did SIP
01:00.08UncleKiwii remember paying some money a few years ago to get the skype channel driver
01:00.29UncleKiwiit was interesting - but turned out to be a waste of time
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02:33.23igcewielingmutters something about zombie werewolves
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02:44.11forgotmynickhello
02:44.56forgotmynickasterisk keeps randomly crashing (process disappears) but nothing is showing in the log files. can someone point me in the direction on how to debug this?
03:00.32drmessanoUncleKiwi: The paid SfA module was built as a Skype "Client"
03:00.59drmessanoIt more or less had special access to the Skype network.. it was a partnership
03:01.14drmessanoWhatsApp has no public API
03:01.27drmessanoSo Asterisk isn't an option
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14:32.44digitalowlHello everybody. Can someone tell me about ooh232 and asterisk?
14:33.49SamotH232 is pretty much dead.
14:33.59SamotBut what are you looking for?
14:35.17digitalowlYeah, but I have avaya 1692 and my partner want connect this shit to asterisk
14:35.31digitalowlI already found gnugk
14:35.43digitalowlAnd register asterisk as gateway
14:35.59SamotAll this for a single phone?
14:36.08SamotTell them to get a new conference phne.
14:36.20digitalowlEm... He want sell this shit to customers
14:37.05SamotUhm.
14:37.17SamotWith Asterisk?
14:37.31digitalowlYeah
14:37.41SamotOn a channel driver that is basically not used by anyone and is the biggest PITA to deal with.
14:37.57SamotAlso a lack of devices that will support it.
14:37.58digitalowlPITA?
14:38.05SamotPain In The Ass
14:38.13digitalowllol, ok
14:38.27SamotSo does he want to sell other H.232 devicecs?
14:38.34SamotOr just a conf. phone?
14:39.00digitalowlHe want sell devices and conf
14:39.14SamotSo why is he picking stuff that is not standard?
14:39.27SamotThat is a pain to get configured and setup not to mention support.
14:39.30SamotSIP
14:40.08digitalowlI don't know, may be he buy some count with a discount
14:41.05digitalowlI think this stuff must be used with avaya or cisco stations?
14:41.46SamotWell they are most likely meant for Avaya PBX systems. Which would also mean that some of the features will most likely *not* work with Asterisk.
14:42.13Samot@file: Is there something up with the wiki?
14:42.25digitalowlBut how about h.323 <-> SIP proxy?
14:42.58SamotYeah, I guess.
14:43.15Samoth323 is a channel driver in Asterisk.
14:43.25SamotBut it makes Asterisk the gateway not the gatekeeper
14:43.34SamotH.323 requires a few things.
14:43.35digitalowlI gound gnugk
14:43.41digitalowl*found
14:43.43SamotIt's why it lost over SIP
14:43.48SamotPart of it.
14:43.58SamotToo many pieces, too complicated.
14:44.35digitalowlTheoretically it's perhaps?
14:45.18SamotWell supporting them is going to be a joy..
14:45.31SamotBecause the amount of people in here that use H.323 is well...
14:45.38SamotProbably you, if you do this.
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15:09.36digitalowlThx
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16:30.10mysthHi! Don`t know if this is right place for asking help, but im having problems with Incrediblepbx(for Rasperry pi 3). I have tried to register Trunk with my provider but Asterisk logs say: https://pastebin.com/hrR5kM88. Can anyone point me at right direction?
16:32.01Samot[2017-07-09 19:05:28] NOTICE[19421] chan_sip.c: -- Registration for '+358xxxx@my.provider.fi' timed out, trying again (Attempt #1136)
16:32.08SamotThat's pretty clear.
16:32.38SamotCall your provider and make sure you are 1) using the right details and 2) they can confirm they see the attempts
16:32.44mysthYeah, but what i can do about it? Why does it time out
16:32.57SamotYou call your provider.
16:33.38mysthI have also Raspberry1. It works fine using Freebpx(the software version is 2014)
16:33.45SamotSo?
16:33.48SamotThey are not the same device.
16:33.58SamotThey are not the same config, as far as I know.
16:34.01mysthYes, i know.
16:34.03SamotEither you have a DNS issue
16:34.07SamotAnd can't resolve that name
16:34.12SamotYou have an Internet issue
16:34.22SamotOr something is not right in the config
16:34.29SamotOr there is an issue with the provider...
16:34.58SamotCan the PBX resolve the FQDN host you are trying to connect to?
16:35.06SamotCan the PBX ping it and get a response?
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16:55.58[TK]D-FenderOr ther is something else wrong in the comm being generated.
16:56.07[TK]D-FenderWe'd have to see the actual attempt in oder to diagnose anything
16:57.48SamotWell I'm still waiting to see if the destination is even resolvable and/or reachable before worrying about sip debugs.
16:58.06Samot1136 timeout attempts....
16:58.15[TK]D-FenderYou shouldn't get a reg attept timeout if it couldn't resolve
16:58.24[TK]D-Fendertaht should mean it sent a packet that never got answered
16:58.38[TK]D-FenderSo you absolutely should go prove the content
16:58.38SamotRight
16:58.42SamotBut its UDP
16:58.48SamotIt doesn't care if the other side is live or not
16:58.50SamotIt just sends
16:59.01[TK]D-Fenderit won't send if there's a DNS issue
16:59.12SamotPlus I doubt there is a provider using "my.provider.fi" as their FQDN.
16:59.19SamotSo waiting for real information.
16:59.46Samotmysth: You want to shed some more light?
16:59.56Samotmysth: Answer those questions?
17:06.56mysthSorry for the delay. I can ping it
17:07.08SamotAlright.
17:07.10SamotSo then..
17:07.17Samotasterisk -r
17:07.20Samotsip set debug on
17:07.26SamotMake a register attempt
17:07.30Samot~pb
17:07.30infobot[pastebin] a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
17:07.37Samot^^^ paste the results there.
17:08.06SamotDo NOT CHANGE THINGS
17:08.17SamotWe need to see the actual, real data being sent.
17:08.43mysthAllright
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17:14.27mysthHere it is: http://pastebin.ca/3840655
17:16.09SamotThe PBX isn't behind NAT?
17:18.07mysthIt is, i have Asus AC66 router and NAT is enabled.
17:18.40SamotThis a new setup or was this working and then this started happening?
17:18.54mysthThis is a new setup.
17:19.18SamotOK, so either they are not getting your initial REGISTER request....
17:19.50SamotOr they are and the reply is lost on the way back to the PBX. Which would point to firewall/NAT rules.
17:20.06drmessanomysth: Is this stock firmware on the AC66?
17:20.15mysthNo, Merlin fw.
17:20.22SamotOh, of course you would know this router.
17:20.26Samotnerd.
17:20.33SamotROUTER GEEK
17:20.59drmessanomysth: Is SIP ALG disabled?
17:21.46mysthNo, it is set "Enabled + NAT helper"
17:21.50drmessanoSamot: I wish I didt
17:21.55drmessanomysth: Disable it
17:22.57mysthOk.
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17:24.17drmessanoSamot: I really like ASUS routers
17:24.23drmessanoSamot: They make it easy to sell an MT
17:24.48Samotrofl.
17:29.26mysthIt is now disabled, but no change. Do i need to boot my router or Raspberry?
17:29.59drmessanoYou didnt reboot?
17:30.11mysthNot yet, no.
17:30.16drmessanoI think you have to in order to disable SIP ALG
17:30.29mysthOk, i will boot it now
17:34.16mysthDone
17:34.33mysthBut still the same :(
17:35.07mysthError code is different, it`s now 1938
17:35.11SamotShow it
17:35.14SamotSame as before.
17:35.19mysthAllright
17:38.59mysthhttps://pastebin.ca/3840666  Here is the new one
17:40.25drmessanoDo you have another router?
17:41.21mysthI can get one from work if i dont find my old one.
17:42.47SamotI would take the 5 minutes to confirm with the provider they are actually getting your requests.
17:43.35mysthI can call them tomorrow.
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19:18.01SamotAnyone else having an issue with the Asterisk Wiki?
19:22.44drmessanoSamot: Are you behind a NAT?
19:23.13drmessanoSamot: It wont load for me
19:23.21SamotOK.
19:23.25SamotJust checking.
19:23.34drmessanoI even rebooted my ASUS
19:23.51SamotDo you have SIP Helpers enabled?
19:24.02SamotAnd is your Outbound NAT right?
19:24.16drmessanoRIght, my double Outbound NAT
19:24.42drmessanoWHICH WORKED FINE BEFORE WITH SOFT FONE
19:24.53drmessanoZOIPER WORK FINE
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19:37.16DovidAsterisk wiki is down?
20:16.59drmessanoYes
20:20.16lorsungcudrmessano: is asterisk wiki down
20:20.18lorsungcui cant access ti
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20:51.08fileit's back up now, Java be Java
20:51.55drmessanoThanks file
20:52.05drmessanoI almost had to look up something
20:52.21drmessanoand almost had to resort to voip-info
20:52.29drmessanoWhich would have meant rolling back to 1.2
20:52.37drmessanoand I didn't want to do that
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20:56.34filethere was no 1.2, it was all a dream
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21:32.37drmessanofile: I didn't hate 1.2.. Skipping 1.4 is ok though
21:33.09drmessanofile: My therapist has a teddy bear with 1.4 written on it, so I can show him on the bear where 1.4 touched me
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