IRC log for #asterisk on 20170126

00:00.12apb1963so take it out of the hosts file?
00:00.22[TK]D-Fenderor prepare an extra resolver, etc
00:00.35[TK]D-Fenderbut yeah, removing is a solid option unless you seriously need that for something else
00:00.42[TK]D-Fenderat which point... get ANOTHER DDNS name
00:00.42apb1963I'm not sure if I do or not
00:00.46[TK]D-Fenderso * has something real
00:00.55apb1963wait.. what?
00:00.58[TK]D-FenderYOUR server... you should know why you do this stuff :p
00:01.23apb1963meh... I set stuff up, it works... I leave it alone.. I get older.
00:01.29apb1963I forget.
00:03.03apb1963can you give me some more info on "another ddns name" ?  Are you saying point a second DNS at the same dynamic IP?
00:03.17apb1963err... FQDN?
00:03.51apb1963just in case I remember to do it some day :)
00:05.28apb1963yeah I don't know about the hosts file...  it has 127.0.1.1 as my FQDN ... localhost isn't the same at 127.0.0.1 so I'm not sure it's not being used for something.  I don't think I set that up, that's installation default I'm reasonably sure.
00:07.24apb1963[TK]D-Fender, look at the second answer here in reference to GNOME.  This is why my machine/network is are always at odds with asterisk.
00:07.31apb1963http://unix.stackexchange.com/questions/11844/etc-hosts-for-debian
00:09.09*** part/#asterisk deweydb_ (~deweydb@S0106689e199caaf4.no.shawcable.net)
00:11.30[TK]D-Fenderapb1963> can you give me some more info on "another ddns name" ?  Are you saying point a second DNS at the same dynamic IP? <- yes
00:11.57[TK]D-Fenderthat machine name is fine
00:12.06[TK]D-Fenderits the EXTERNAL one you should make sure is not in there
00:12.19apb1963well that IS the external one
00:12.21[TK]D-Fenderbecause no interna service should point to your FQDN name
00:12.46apb1963Please read the link I mention above.  GNOME uses it.
00:13.01[TK]D-FenderGnome shouldn't know anything about you having a domain name
00:13.08[TK]D-Fenderyour MACHINE NAME should not match <-
00:13.25[TK]D-FenderIf you did that ... go slap yourself for a SECOND shoo-on-sight-offense :p
00:13.29drmessanoYou are seriously misunderstanding FQDN from an external and internal context
00:14.20drmessanoserver01.internaldomain.com is not the same as www.mydomain.com.  Hosts is referring to your machine name, which may exist in an INTERNAL DOMAIN
00:15.00apb1963From the link I mentioned:
00:15.04apb1963In your case, because you don't have a permanent IP address, you can also have
00:15.04apb1963127.0.1.1    machinename.domain machinename
00:15.04apb1963This line seems to be required for some applications like GNOME, but it might actually cause problems with other applications!
00:15.23drmessanomachinename.domain is NOT THE EXTERNAL ADDRESS
00:16.01apb1963Now, if that's not GNOME knowing something about having a domain name, then obviously I don't understand what that is saying.
00:16.08drmessanoYoure not
00:16.14drmessanoNot at all understanding
00:16.45drmessanopbx01.local is an example of a FQDN on an INTERNAL domain
00:17.01drmessanoIts NEVER your internet hostname
00:18.05apb1963My hosts file contains, by default - the following:
00:18.09apb1963127.0.0.1       localhost
00:18.09apb1963127.0.1.1       blue.publicserviceclub.com      blue
00:18.20[TK]D-Fender#2 = fail
00:18.22drmessanoand thats wrong
00:18.40apb1963I didn't set it that way, the installation did.
00:19.01[TK]D-FenderI know I never tell my servers that kind of info.
00:19.04apb1963That is the default hosts file as setup by the install scripts.
00:19.06drmessanoYou dont give a box the same FQDN as the outside
00:19.06[TK]D-Fendercomment it out
00:19.22[TK]D-Fenderbrb
00:19.56apb1963If you read the link I provided above, you should understand why they did that.
00:20.09drmessanoIm not the one having a hard time understanding
00:20.13apb1963It is not wrong for GNOME, if I'm understanding the link correctly.
00:20.19drmessanoIt IS wrong
00:20.44apb1963Then why did the distribution maintainers decide it was right?
00:20.57drmessanoThey didnt
00:21.09drmessanoYou misinterpreted the machinename.domain
00:21.10apb1963Then why is it set that way in the file?
00:21.15drmessanoThey dont want the EXTERNAL HOST NAME
00:21.38apb1963They didn't ask for both an internal and external, they asked for one host name.
00:21.40drmessanoYou've obviously no concept of an INTERNAL DOMAIN
00:21.44drmessanoRIGHT
00:21.59drmessanoYour BOX is not blue.publicserviceclub.com
00:22.22drmessanoYou TOLD the installer that
00:23.30drmessanoAs I said above
00:23.34drmessanoand I will say it again
00:24.01apb1963fair enough... so I should have told it what, precisely.  So I can understand and get it right.
00:24.14drmessanomachinename.domain is expecting something like pbx01.myinternaldomain
00:24.23drmessanoNot your internet hostname
00:24.53apb1963"something like".  That's what I'm not understanding.  What exactly?  Something I make up and it doesn't matter?  Or is it relevant to something related?
00:24.59drmessanoANYTHING
00:25.02apb1963ok
00:25.03drmessanoIts an internal domain
00:25.07drmessanoblue.local
00:25.09drmessanoblue.internal
00:25.16drmessanopbx01.blue.local
00:25.29drmessanoAnything BUT the internet hostname
00:25.59*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
00:26.29apb1963cool.  thank you.   So "127.0.1.1 blue.internal" will do it?
00:26.34drmessanoSure
00:27.06SamotI wish free Google Voice would die.
00:27.17drmessanoSoon
00:27.24drmessanoThey are killing hangouts
00:27.28apb1963odd... I had thought 127.0.0.1 was all that was needed.
00:27.30SamotOh yeah?
00:27.31drmessanoSo XMPP is surely gone next
00:27.42drmessanoapb1963: Its not a matter of needed
00:27.55apb1963?
00:28.01drmessanoYou told the PBX to resolve blue.publicserviceclub.com to 127.0.1.1
00:28.08drmessanoSo its using it in invites
00:28.13drmessanoThats not gonna work
00:29.06apb1963click.  Got it.  thank you
00:29.48SamotI don't know..they've been saying Hangouts and GV were going to die for almost a year now.
00:30.01drmessanoSamot: Negative
00:30.02apb1963hmm... restart asterisk?
00:30.08drmessanoapb1963: Yes
00:30.16drmessanoSamot: Google Voice was never going to die
00:30.25drmessanoSamot: They were killing Talk
00:30.48apb1963Samot, you sell phone services?
00:30.51drmessanoSamot: With the new chat apps, they decided to kill Hangouts
00:31.07SamotYeah, I saw something about that last year.
00:31.21SamotBut I thought they said Hangouts was going to stay despite the new apps.
00:31.28drmessanoNope
00:31.37SamotI know they dump some On Air thing to YouTube Live.
00:31.40drmessanoThey've also revamped GV
00:31.44drmessanoUpdated apps for it
00:31.47SamotSince when?
00:31.56drmessanoIts rolling out this week
00:32.02Samotapb1963: Yes.
00:32.03drmessanoNew features coming
00:32.20apb1963Samot, now I understand your dislike of GV :)
00:32.30SamotNo.
00:32.38drmessanoThats lame
00:32.41SamotThat absolutely has nothing to do with it.
00:32.56drmessanoYou can apologize to my friend
00:33.05apb1963woohoo!
00:33.17apb1963It's working
00:33.20apb1963Thanks!!
00:34.47apb1963Yeah, that machine.internaldomain vs external domain... I've never seen that documented anywhere... or explained well enough to understand it.
00:35.28apb1963So I appreciate you taking the time to educate me. drmessano
00:35.34apb1963Thanks again!  Peace out.
00:35.43apb1963(as they say nowadays :)
00:37.30Samotapb1963: It's well documented.
00:37.42SamotLike basics of linux and DNS.
00:39.17*** part/#asterisk kharwell (kharwell@nat/digium/x-awouqeflyffnzubi)
00:39.56Samotdrmessano: He still hasn't apologized and like my feels are all hurting.
00:45.04SamotOoh DigitalOcean is bring in load balancing..
00:45.12Samots/bring/bringin/
00:45.54SamotAPI based as well..
00:48.03fileSamot: ECMP based using BGP interests me... more
00:50.41SamotThere's not much info on how they are doing it. It was just a 2017 roadmap announcement.
00:50.49SamotSo I will be interested to see how they do it.
00:52.00fileit wouldn't surprise me if it's actually an nginx based image in a droplet that is driven based on data from their backend, with the API they mention
00:56.22*** join/#asterisk chendy (~alexc@183.12.65.129)
01:11.01dan_jI'm suddenly getting 'Exceptionally long voice queue length queuing to' when asterisk tries to playback the voicemail greeting. I've tried restarting asterisk but that doesn't help.
01:11.22dan_jI've not changed anything. It's suddenly started happening.
01:11.31dan_jWhat does it mean?
01:13.05dan_jthe debug shows this at the same time 'ast_find_ourip: Not an IPv4 nor IPv6 address, cannot get port'
01:13.31SamotShow the call and debug
01:15.39dan_jHmm. It's happening on both asterisk servers that are in that pacemaker cluster
01:15.48dan_jOne moment.
01:18.07dan_jhttps://www.irccloud.com/pastebin/bWuVlHQK/
01:19.18dan_jAll I did was re-record a greeting, but it wasn't from within app_voicemail. It was recorded via my own script and saved to the voicemail realtime db
01:19.55dan_jI've opened the recording from the db and it's in the correct format.
01:20.31SamotNo...
01:20.39SamotI don't want a core debug.
01:20.50SamotI want a verbose call with a SIP debg.
01:20.56SamotSo pjsip set logger on
01:20.58dan_jAnd I can't seem to hang up that channel now. It's just stuck in 'core show channels' even if i request a hangup.
01:21.05Samotcore set debug 0
01:21.08dan_joh. ok. onesec
01:21.09Samotcore set verbose 10
01:21.13Samotpjsip set logger on
01:21.14dan_ji thought you meant core debug
01:21.20SamotNo.
01:21.26dan_jone moment
01:23.10dan_jgonna be difficult to get a clean sip debug. the call is originating from my carrier and they use a range of ips. there are 50 endpoints connected, so if i do set logger on, i'll get all their packets too
01:23.17dan_jwill try to get it as clean as possible.
01:23.48SamotThen just a verbose call for now.
01:25.39dan_jhttps://www.irccloud.com/pastebin/nQnTQ4cZ/
01:25.47dan_jSorry, only just saw what you wrote.
01:27.24dan_jhttps://www.irccloud.com/pastebin/j9NIajMX/
01:27.30dan_jThats just the verbose
01:27.34*** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica)
01:28.10dan_jJust going to check my other cluster that uses the same voicemail database
01:28.15klowon chan_sip "sip show channelstats"  , how does asterisk know % of packets lost on  send
01:29.06SamotWell something is happening.
01:29.11SamotThis looks like a deadlock issue.
01:29.13dan_jThe other cluster seems fine
01:29.34dan_jExactly the same dialplan and voicemail db
01:29.41SamotYou're creating a local channel to call the voicemail.
01:30.15SamotThat error generally means there's nothing servicing the channel, from what I remember.
01:30.49dan_jYes. thats what i saw too. but it's weird because its suddenly started happening.
01:30.56dan_jand a reboot doesnt seem to shift it.
01:31.08dan_jI'm going to re-record the greeting from within app_voicemail and see if the issue continues.
01:40.08dan_jHmm. Seems to be fine when recorded via app_voicemail. This is the first time i'm using my recording code with v13. Works fine with v11. I wonder if something has changed that makes it not compatible.
01:40.17dan_jCant imagine what though.
01:44.34dan_jThe file plays back fine before it gets saved to the db. And then when app_voicemail tries to play it, it gets stuck.
01:50.27dan_jGonna have to try this again tomorrow. Thanks for the help.
02:11.03*** join/#asterisk sarsaeol (~sarsaeol@unaffiliated/sarsaeol)
02:13.23sarsaeoli am taking over an asterisk system that used to send voicemail notifications through MS exchange, wondering why the imapuser might look like a path, is this domain info? ex: 3005 => 1456369,User Name,,,imapuser=company\asteriskvmail\username|imappassword=password
02:14.00sarsaeoli've only ever personally seen the imapuser be something like "user@gmail.com" or similar
02:14.25SamotCan't say.
02:14.38SamotIMAP based voicemail is gross.
02:15.17SamotFactor in that locals like some fileshare thing MS would do.
02:15.23SamotEven more gross.
02:15.28sarsaeoloh god
02:16.35SamotWell I'm not sure.
02:17.09SamotI mean if I *had* to do IMAP for voicemail, Exchange would not me on any list of choices as the IMAP server.
02:17.16Samots/me/be/
02:17.31sarsaeollol gj bot
02:17.52SamotBut, that could be the domain info.
02:17.54SamotBecause Ms.
02:18.20SamotWell domain\user details.
02:19.55SamotGuessing that multiple users check the same IMAP box...
02:20.08SamotSo they can see what VMs are there and if they've been read, etc.
02:23.14sarsaeolokay, looks like i need to read up on IMAP VM storage then
02:23.48sarsaeoli had assumed this was just attaching a wav and sending a notification without real integration with the voicemail box, what a can of worms i've opened today
02:24.08SamotNo.
02:24.14SamotThis is actual IMAP based storage.
02:24.39sarsaeolyup, it's becoming more clear with each passing second =\
02:24.44sarsaeolsighs
02:25.19sarsaeolin any case, cheers for clearing that up Samot, luckily they are moving off of exchange, so perhaps that will clear some things up in the config
02:25.34SamotNot really.
02:25.37sarsaeollol
02:25.43SamotExchange is just the IMAP server.
02:26.01SamotThe question is, do they want to change from IMAP storage?
02:26.18SamotOtherwise you're just setting up a new IMAP server to store voicemails at.
02:27.10SamotThe benefits are the voicemails appears as emails in their client.
02:27.34SamotAnd those voicemails are treated like emails. The other users can see a message was read or moved to another folder..etc.
02:28.46SamotChanging storage methods changes their entire method of managing and accessing their voicemails.
02:29.14SamotSo yeah, this is a big, ugly can of worms.
02:29.20sarsaeol\o/
02:29.53SamotThey could have 20 users managing one voicemail account via IMAP.
02:30.24SamotAll used to managing their VM like an email.
02:30.46SamotSo. Many. Will. Whine.
02:35.52SamotSucks to be you.
02:36.09sarsaeolhahahaha i see that now!
02:36.40sarsaeolthey came in out of the blue, never worked with us before, the boss puts it on my desk like "this is easy, they want it done by friday"
02:37.39sarsaeolalthough it does seem like every mailbox in the relevant context is tied to only the individual user, no perhaps no shared mailboxes etc.
02:37.56sarsaeolbut yeah, replicating this on a new IMAP server is gonna be a pain in my ass
02:39.13SamotWhat are they moving to?
02:39.20SamotFor an email server?
02:39.25sarsaeolbrace yourself
02:39.30sarsaeoloffice 365
02:39.38SamotSo total MS heads.
02:39.43sarsaeolso, out of the frying pan into the fire
02:39.52sarsaeolyeah aside from this one asterisk box i guess =D
02:39.54SamotYou may want to inform your boss of all this.
02:39.59sarsaeoloh i will for sure
02:40.06SamotAnd tell him Friday ain't happening.
02:40.20SamotOne, you'll need a test box.
02:40.29sarsaeolhe quoted them like 2 hours (out minimum) because he thought it was just changing the smtp server used to send email notifications
02:40.29SamotSo you can test imap storage setup and stuff.
02:40.32SamotBecause I bet you..
02:40.34SamotRight now..
02:40.45SamotThey 100% expect their voicemail to move with their email.
02:42.03SamotThey are doing nothing more than converting from an on-premise Exchange to basically a hosted per seat version.
02:43.16Samot"We get our voicemails in email."
02:43.27sarsaeolthat's all they said!
02:43.35sarsaeolhow'd you know!? ;)
02:43.44SamotShould be followed by "You mean email notices?"
02:44.04sarsaeoli believe that /was/ the follow up, but the person on the other line didn't know the real difference
02:44.21sarsaeoland so just said 'yes' to 'are they just email notifications'
02:44.26sarsaeolwhen really...
02:44.40SamotThe email lives in IMAP
02:45.24sarsaeolright, well at least I understand the difference now and will learn something
02:45.36sarsaeoleven if we just drop them like a hot potato =P
02:46.31SamotWell it's more than two hours, that's for sure.
02:46.51SamotLike I said, you'll need to test the IMAP connection, etc with Office 365.
02:46.59SamotSo you'll need a box that can do that.
02:47.19SamotCan't do it on theirs, you'll break their current voicemail.
02:47.39sarsaeolgives the client to Samot and runs away
02:47.45SamotOh hell no.
02:47.55sarsaeolno take-backs!
02:48.01SamotI'd never would have gotten this far with them.
02:48.44sarsaeolyeah this tends to happen when sales doesn't talk to noc
02:49.12SamotThat's why there should be a "Sales Engineer"
02:49.31sarsaeolno argument from me
02:49.43sarsaeoli can set up a server and replicate their setup pretty easily tomorrow, that will be the easy part
02:49.50SamotI have a buddy that works at Cloudera..
02:49.54SamotHe's a Sales Engineer..
02:50.03SamotHis job is to go where sales guys go.
02:50.11SamotBe in all the meetings.
02:50.22SamotAnd shake his head no while the sales guy says "yes".
02:50.49igcewielingremembers a series of Dilbert comics
02:51.43SamotWhen I joined the CLEC they didn't have a sales engineer and they kept screwing up their IP based sales for voice.
02:51.51SamotBecause of stuff like this.
02:52.22SamotBut like after contracts were signed and the balls was rolling..so a complete waste.
02:52.35SamotOr even getting to turn up to find out stuff isnt right.
02:52.53sarsaeolsounds like my daily life tbh
02:53.04Samot$1,200/month contracts down the drain..
02:53.04sarsaeolthere's always some new clusterfuck around every corner
02:53.18SamotWell I stopped that.
02:54.33sarsaeoli make valiant attempts to instill some critical thinking in the people who make these decisions, it just doesn't always stick when they are faced with desperate people willing to pay
02:54.52SamotRevenue isn't revenue if it costs more to do it.
02:55.05SamotIf it's $125/hour and its two hours...
02:55.26SamotAnd you have to spend three hours plus spin up a system for testing, etc.
02:55.38SamotNow that sale put you in the red.
02:56.04SamotNot only for that sale but it was three hours of time you could have been doing revenue generating work.
02:56.25sarsaeoli mean, we wouldn't eat the cost, can't afford to. we'll let them know the scope has changed. luckilu no support contract was signed, this was jsut a prelim estimate
02:56.51sarsaeolbut your point is well made, it's exactly the reason we need some education in sales
02:57.16SamotWell someone in engineering should have been bitching by now.
02:58.28sarsaeollol there's literally no one but me here
02:58.36sarsaeolbut in the morning!
02:58.41SamotOf course, I did piss a lot of people off with it.
02:58.42sarsaeoli will raise hell =]
02:58.54SamotBecause everyone worked at the same time..
02:59.25SamotSo if there were five pieces to the job, five people would work on their own..
02:59.30SamotNo structure.
03:00.00SamotAnd they would step on each other, change what the others did or just set stuff up without all the details..
03:00.13SamotBecause someone had to give them and hadn't.
03:00.40SamotA system was put into place that wouldn't let steps be done unless other steps/processes were completed.
03:00.55SamotSo it started to show who was lazy and half assing their work.
03:03.29SamotThank god I left. I will never go back to a large company again.
03:24.32*** join/#asterisk fstd_ (~fstd@unaffiliated/fisted)
03:26.56sarsaeolthanks for all the insight Samot o/
03:27.00sarsaeolmuch obliged
03:27.17*** part/#asterisk sarsaeol (~sarsaeol@unaffiliated/sarsaeol)
03:35.36*** join/#asterisk clopez_ (~tau@neutrino.es)
03:39.04*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
03:43.54vader-can this (http://www.kaplansoft.com/teksip/) be used to connect a SIP Trunk to and then connect a SIP Device just for a simple call tansfer?
04:05.34*** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net)
05:03.33*** join/#asterisk putnopvut_ (putnopvut@asterisk/master-of-queues/mmichelson)
05:03.33*** mode/#asterisk [+o putnopvut_] by ChanServ
05:11.02*** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux)
05:13.19drmessanovader-: Why wouldn't you use Asterisk for this?
05:13.31drmessanoIt kinda absolutely does the same thing
05:13.38drmessanoIt's a B2BUA
05:14.14vader-just trying something quick and simple
05:14.34vader-im breaking a number off the sip trunk and redirecting it to this so i can use this phone
05:14.40drmessanoASTERISK is quick and simple
05:14.45vader-kinda have it semi working
05:14.51vader-i can't get any inbound calls
05:14.54vader-working
05:15.02vader-outbound is working but i can't hear the remote caller
05:15.12drmessanoI'm sure they have excellent support
05:15.21vader-2talk.com is the sip trunk provider
05:21.07SamotHe did this in #freepbx.
05:21.28SamotDude, if you're going to use MS based voice products you need to find a room/site for it.
05:23.08drmessanoOh lovely
05:24.11SamotHis reason: No other IP telephony rooms, really.
05:26.23drmessanoYeah, no
05:26.37drmessanoIf you're going to use commercial software
05:26.40drmessanoThey have support
05:26.42drmessanoGo call them
05:27.36drmessanoFor $219 you also get support with TekSIP
06:00.12*** join/#asterisk fbnts (~fbnts@s099.spireinns.co.uk)
06:19.15*** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh)
06:21.10*** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66)
06:29.05vader-ya i was trying to get away with using their free version since all i am trying to do proxy 1 trunk to 1 device
06:29.15vader-thought this would be a good use case for their free version of the app
06:29.49drmessanoAsterisk works too
06:33.41SamotBut why ask help in a place that's nothing near what you are doing?
06:33.51SamotFirst it was Skype SFB.
06:34.11SamotNow it's some Windows SIP proxy.
06:57.17*** join/#asterisk Pasha (~Cory@unaffiliated/cory)
07:02.08*** join/#asterisk bof22 (~Thunderbi@185.13.183.107)
07:13.42vader-Samot, people in here and in ##freepbx deal with a wide variety of systems. Someone may have run across something similar or may have insight... I don't see where my question caused so much pain.
07:14.30SamotYou ask questions in the form of "will this work?" or "can i do this with X?"
07:14.55SamotAnd yes, people *could* have some experience with it..
07:15.07SamotBut highly unlikely...
07:15.34SamotWe helped with SFB..
07:15.42SamotBut we were actually Googling things.
07:16.39SamotI get people asking, every once in awhile, about other things...
07:16.57SamotCould be in relation to Asterisk or their use of Asterisk, such as phones or Kamailio...
07:17.09*** join/#asterisk jkroon (~jkroon@dustpuppy.is.co.za)
07:17.17SamotBut you have continually done this.
07:17.42SamotYou don't ask the occasional non-Asterisk/FreePBX question. Its what you ask the most of.
07:17.49SamotAnd ask for help.
07:17.54Samot"How can I do this?"
07:19.05Samot12:26:34 AM <drmessano> If you're going to use commercial software
07:19.14Samot12:26:38 AM <drmessano> They have support
07:19.36drmessanoIt's one thing if you're using it WITH Asterisk
07:19.37SamotIt's kind of a thing people look at when they opt for commercial vs. open source.
07:19.41drmessanoWe all love a challenge
07:19.42SamotYes.
07:19.48SamotThe SIP proxy in front of Asterisk..
07:19.53SamotWe'd be up for that.
07:19.56drmessanoBut asking about unrelated software is kinda uncool
07:20.04SamotThe SIP proxy in front of SFB..
07:20.05SamotNo.
07:20.57SamotNot to mention..
07:21.09SamotGoing into multiple rooms and blasting them all with the questions.
07:21.14drmessanoWhat I dont think you understand
07:21.15SamotLike a buckshot..
07:21.26drmessanoTheres ASTERISK
07:21.30drmessanoWhich DOES THIS
07:21.35*** join/#asterisk WIMPy (~wimpy@x4dbaec9c.dyn.telefonica.de)
07:22.05drmessanoSo its not like you're asking about something unrelated.. you're asking in #ASTERISK about another piece of software that does EXACTLY the thing that ASTERISK does and *WHY WE ARE HERE*
07:22.24drmessanoThats like going to #MySQL and asking Postgres questions
07:22.38drmessanoif you don't want to use Asterisk, why ask in #Asterisk?
07:22.40SamotWell it's the only database room.
07:22.50drmessanoJUST USE FUCKING ASTERISK
07:23.28drmessanoYou do realize you can install it and only configure like 2 SIP peers and be done with it?
07:23.38SamotOr read the manuals/docs/support data.
07:23.47drmessanoBasic dialplan.. Take all calls on Peer 1 and send them to Peer 2
07:23.50drmessanoDone, handled
07:24.00SamotOr you can install *gasp* FreePBX.
07:24.11SamotAnd be done even quicker, cuz like GUI.
07:24.13drmessano"I only need something Simple"
07:24.20drmessanoAsterisk is literally something *simple*
07:24.36SamotMicrosoft voice servers and MS based SIP proxies..
07:24.38SamotNot simple.
07:24.58SamotSFB is taking a butcher knife to something when you needed a scapel.
07:25.06SamotSFB is taking a butcher knife to something when you needed a scalpel.
07:25.29drmessanoBut anywho.. this is just equine necromancing
07:25.46SamotOh.
07:25.48SamotDead Horse.
07:25.53drmessanoIf you dont want to use Asterisk, stop asking for help
07:25.56SamotI'm literally listening to it right now.
07:26.31SamotUse Your Illusion I
07:27.00SamotNo body understands quite while we're here...searching for answers that never appear...
07:27.23SamotAnd when she said, she like wrecked my car. I didn't know what to do!
07:28.39SamotBut yeah, what he said.
07:28.51drmessano6 people shot in Chicago at the memorial for a shooting victim.  Wounded include the victims mother and a 12 year old girl
07:29.06drmessanoChicago needs to just be walled off
07:29.10SamotI just found it funny I was listening to Dead Horse when you made the equine comment.
07:29.13SamotNo shit.
07:29.38Samot#ThanksChicagoForMakingDetroitLookGreatAgain
07:29.49drmessano#SoTruthy
07:30.18SamotBecause wasn't the line of thought a while back that Detroit needed to be walled off....
07:30.21drmessanoGive it a few more weeks and there wont be anyone left
07:30.23SamotOr given to Canada..
07:30.42drmessanoYeah, Detroit was like first in line for forced secession
07:30.58SamotAnd for the record..
07:31.08SamotThat fscking Journey song is about Windsor.
07:31.18SamotTHERE'S NO SOUTH DETROIT.
07:31.24drmessanolol
07:31.41drmessanoI want to move to Ontario
07:31.49SamotWindsor just happens to be south of Detroit.
07:32.12SamotIt's one of the few spots that people have to go South to get to Canada from the lower 48.
07:32.15drmessanoI keep seeing Ontario license plates everytime I travel
07:32.26SamotHell even Alaskan's have to go east.
07:32.39SamotAnd north..
07:32.58SamotDetroit, you go south to get to the Great White North.
07:32.59drmessanoSo Ontario is like the Georgia of Canada
07:33.42SamotYeah.
07:33.59drmessanoAre they 25 years behind?
07:34.24SamotThey don't have the little metal flaps on gas pumps.
07:34.30SamotSo you don't have to hold the handle.
07:35.10SamotStores just got self checkout 2016
07:35.22SamotLike some are still converting.
07:36.09SamotMy wife's cell company just got LTE
07:36.14SamotAnd will be 4G by end of 2017.
07:36.45drmessanoWOw
07:36.48SamotYeah.
07:37.02SamotWhen I went there last year to get service..
07:37.10SamotThey said they couldn't use my LG Vista.
07:37.16SamotBecause they were only 3G
07:37.29SamotThey said, "Yeah by Q3 or so 2017 we'll have 4G"
07:37.44SamotAnd I said "Yeah but that's about the time 5G is going to start rolling"
07:38.17SamotSo I think their LTE/4G is hand me downs.
07:38.34drmessanoThats not uncommon
07:38.46drmessanoSouth America is grateful for all the EDGE gear
07:38.57SamotWell its one of those pay as you go places.
07:39.05drmessanoOh right
07:39.09SamotSo they use Rogers
07:39.14SamotAnd T-Mobile
07:39.21drmessanoWait until Canada gets the iPhone
07:39.25SamotHaha.
07:39.33drmessanoIt has apps
07:39.36drmessanoPretty rad
07:39.41SamotOh man..
07:40.08SamotMy wife's Galaxy Notebook II is having antenna issues.
07:40.17SamotSo she has to replace it.
07:40.21SamotAnd she's not happy.
07:40.23SamotWhy?
07:40.58Samot"I'll have to start Bubble Bobble all over again. I'm at a REALLY high level. It took a long time to get there. I have to re-do ALL OF IT"
07:42.17drmessano:(
07:52.46*** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net)
08:10.18*** join/#asterisk pchero_work (~pchero@2a00:c80:1072::c96)
08:10.57*** join/#asterisk tuxian (~tuxian@igilmour.plus.com)
08:16.03*** join/#asterisk gusto (~gusto@2001:4c50:62e:443:ee34:53bd:4cbf:22a1)
08:17.26*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
08:19.52*** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be)
08:21.33*** join/#asterisk chendy (~alexc@121.34.146.10)
08:26.06*** join/#asterisk tuxian (~tuxian@igilmour.plus.com)
08:26.11*** join/#asterisk hehol (~hehol@gatekeeper.loca.net)
08:44.38*** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson)
08:44.38*** mode/#asterisk [+o putnopvut] by ChanServ
08:56.21TandyUKlol Samot, is she not logged into google play?
08:59.33*** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net)
09:15.53*** join/#asterisk Tiffon (~name@unaffiliated/tiff0n)
09:32.31*** join/#asterisk ace_me (~IceChat9@unaffiliated/ace-me/x-814638)
09:53.02*** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta)
09:54.00*** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta)
09:58.09*** join/#asterisk TandyUK (~admin@87.252.44.195)
10:06.50*** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es)
10:21.44*** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc)
10:25.23*** join/#asterisk mbecroft (~user@ak2.becroft.co.nz)
10:28.22*** join/#asterisk tuxian (~tuxian@194.12.3.78)
10:30.39*** join/#asterisk u0m3_ (~u0m3@86.127.133.199)
10:31.07*** join/#asterisk fbnts (~fbnts@s099.spireinns.co.uk)
10:43.37*** join/#asterisk chendy (~alexc@121.34.146.10)
10:46.35*** join/#asterisk sekil (~sekil@cable-89-216-194-185.dynamic.sbb.rs)
11:26.39jkroonusing asterisk realtime - how can I set multiple setvar statements?
12:07.35*** join/#asterisk brad_mssw (~brad@66.129.88.50)
12:09.40*** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66)
12:23.24*** join/#asterisk chendy (~alexc@121.34.146.10)
12:53.48*** join/#asterisk libardi (~libardi@187.64.235.241)
13:02.34*** join/#asterisk axp (~axp@80.121.172.38)
13:04.42axphi
13:06.39axpi do have a running asterisk installation, however, if the network goes down, Asterisk will not take any calls, is there a way that asterisk checks if the phones are connected? maybe the reinvite parameter?
13:09.49WIMPyTake? Or drop? Not sure what the question is.
13:12.41*** join/#asterisk agent_white (~agent_whi@unaffiliated/agent-white/x-6197888)
13:23.58axpWIMPy, the problem is if i restart e.g. my network switch all SIP phone will not ring
13:29.17*** join/#asterisk [TK]D-Fender (~joe@216.191.106.165)
13:34.11[TK]D-Fenderaxp, get an internal DNS server
13:34.24[TK]D-Fender* can lock up waiting for DNS requeets to resolve
13:35.33axpi do have an internal DNS
13:40.51[TK]D-Fenderis tthat what ou have * pointed to?
13:42.49*** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n)
13:42.49*** mode/#asterisk [+o cresl1n] by ChanServ
13:45.17WIMPyThat doesn't prevent Asterisk from doing strange things.
13:45.56WIMPyI have to admit it was quite painfull, but I finally got the Astribank working on the Raspberry Pi. \o/
13:48.09tzafrirWIMPy, you need dahdi-tools 2.11
13:48.40tzafrirThat is: using libusbx / libusb1 instead of the old libusb 0.1
13:49.23tzafrirwonders who will be at FOSDEM
13:50.08WIMPyThat was there already.
13:50.31tzafrirAny other issues?
13:51.34WIMPyMy last big issue was with system includes and Asterisk itself.
13:52.10WIMPyBut getting DAHDI on that thing was a bit of a journey.
14:04.59*** join/#asterisk rwb (~Thunderbi@204.13.43.166)
14:16.37WIMPyLooks like th RPi can manage up to 5 channels MOH. If there is no networkl traffic.
14:16.37*** join/#asterisk theGoat (~textual@pool-72-94-172-191.phlapa.fios.verizon.net)
14:17.00*** join/#asterisk chendy (~alexc@121.34.146.10)
14:21.15dan_jHi. I've got an agi script which allows users to update their voicemail greeting without accessing the voicemail system. the messages are stored in mysql, but recently asterisk has been complaining about the format. In v11, i get warnings about the format, and in v13 it just deadlocks.
14:21.19dan_jHeres my agi script
14:21.25dan_jhttps://www.irccloud.com/pastebin/lTyf2Zis/
14:21.41dan_jThe insert is done on line 67
14:21.50dan_jAny ideas whats wrong? It has previously worked.
14:22.15dan_jThe recording is done on  line 77
14:22.51dan_jThe CLI says this when it tries to give it a go
14:22.54dan_jhttps://www.irccloud.com/pastebin/SAWpjyaW/
14:24.52theGoatis the file there, if not, try and touch the file to create it and see what happens
14:27.22*** join/#asterisk brad_mssw (~brad@66.129.88.50)
14:28.45dan_jtheGoat: It is there and it can be previewed on line 58 prior to saving it to the database.
14:29.35dan_jIt works fine before it's saved to the db, but once saved, it can't be replayed by asterisk. But I can download the blob using mysql workbench and play the file fine on my pc.
14:30.17theGoatanything in the mysql logs?
14:30.43*** join/#asterisk kharwell (kharwell@nat/digium/x-fqssxtindnnaycbp)
14:30.43*** mode/#asterisk [+o kharwell] by ChanServ
14:37.48dan_jtheGoat: I'll enable general logging and debug so i can get more info from mysql
14:52.03dan_jtheGoat: Nothing showing in the logs. No errors and looking at the insert, it seems fine.
14:52.37Samotdan_j: wav files have to be done in a certain format.
14:52.54dan_jSamot: they are recorded from asterisk so they are in the correct format already.
14:53.17dan_jthe script i'm using to record is my first pastebin, line 77 is the recording line.
14:53.34dan_jAlthough i've changed the format back to 'wav' instead of 'WAV' since I posted that.
14:53.55SamotWhat happens when you record the voicemail greeting normally?
14:54.02dan_jworks fine.
14:54.20dan_jAnd i've compared the two files and audicity tells me they are the same format.
14:54.47dan_jsomething i'm missing but can't figure out what. it has worked for months and suddenly decided to stop working.
14:55.29theGoatupgraded asterisk or mysql?
14:56.43dan_jActually, it's working on my old v11 boxes. But it isn't working on my new v13 boxes and also a test v11 box I just set up don't work. So maybe something has been updated. I've not updated my old v11 boxes for months.
14:57.01tzafrirWIMPy, we now use rPi2 as a platform. One problem with it is that all hardware interrupts (including USB. USB network, SD) are routed to the first core
14:57.03dan_jI'm running the same v11 version on all my boxes, but my test one is failing
14:58.23dan_jI'm going to upgrade mysql to see if that fixes things.
14:58.29theGoathow about running a packet capture while the db traffic is going on.  or maybe something like selinux or something like that is borking it
14:58.52theGoatlook at the linux audit logs first to see if there is anything
14:58.56WIMPytzafrir: I'm currently on a 1B. And that seems to be maxed out at either 5 dahdi channels or 2 dahdi+2 iax channels.
14:59.37tzafrirAh, OK. I was going to point you to our packages repo (http://updates.xorcom.com/servers/spark/), but that is irrelevant for you, then
15:00.21WIMPyI've got some model 2 here as well. Haven't ever used them, yet.
15:00.56Samotdan_j: Show your voicemail.conf
15:01.34dan_jhttps://www.irccloud.com/pastebin/TEMBDcDU/
15:01.40dan_jFrom my test server
15:05.14SamotSo not using the AGI, it puts the recording in the database table without issue?
15:05.22SamotPlaysback just fine?
15:08.56dan_jYes. In AGI, recording (line 77) and then playing back (line 58) the file is fine.
15:09.26dan_jIt's something about the insert.
15:09.47dan_jI'm just upgrading mysql to match the version that the script works on. I didnt realise that yum installed mysql 5.1
15:12.02SamotWhy does the insert matter?
15:12.21SamotIf you record a voicemail straight through the system and avoid the AGI, it works fine.
15:12.27WIMPytzafrir: Is the dialtone generated on Astribank and not sent over USB?
15:12.28*** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net)
15:12.45SamotIt's still using the database for voicemail either way.
15:12.50SamotSo how is it the insert?
15:12.57SamotOr it being MySQL 5.1?
15:13.51tzafrirWIMPy, if it is generated, it is generated on the host and not on the Astribank.
15:14.14tzafrirWith DAHDI the dialtone is generated at the DAHDI kernel driver, IIRC
15:15.01WIMPytzafrir: Hmm. Then playing file must be the hard part, not the USB stuff.
15:15.37dan_jSamot: Sorry. It was the only thing i could think of that wasn't the same as the working system.
15:15.53tzafrirplaying a wav file? That is mostly cached in the memory already?
15:16.02dan_jSamot: Upgrading didnt help.
15:16.06Samotdan_j: So a verbose debug of you recording the call via the normal voicemail system.
15:16.11Samotasterisk -rvvvvvvvvvvv
15:16.18SamotThats it..
15:16.34dan_jvia the normal voicemail system? I know that works though.
15:16.39SamotI know.
15:16.41SamotShow it.
15:16.46WIMPytzafrir: Yes.
15:16.49dan_jOk. 1sec
15:16.50SamotThen you're going to show one that goes through your AGI.
15:16.51*** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta)
15:16.55SamotAnd we're going to compare.
15:16.57SamotBut remember..
15:17.03SamotNormal first..
15:17.14SamotThen AGI based with "agi set debug on"
15:17.20WIMPyAnd yes, I have them as wav, not alaw.
15:19.03*** join/#asterisk scgm11_ (~textual@r179-24-146-130.dialup.adsl.anteldata.net.uy)
15:25.36dan_jSamot: Verbose of normal system. Doesn't offer any feedback on what mysql is doing though.
15:25.52SamotThat's fine.
15:26.24dan_jhttps://www.irccloud.com/pastebin/s2SDME68/
15:27.26SamotOK.
15:27.35SamotNow do it with the AGI metho.
15:27.38SamotNow do it with the AGI method
15:27.43Samotwith: agi set debug on
15:28.40dan_jhttps://www.irccloud.com/pastebin/QoJDG67p/
15:29.58dan_jI played back the recording on line 253 prior to saving it between line 259 and 260.
15:30.36dan_jI've been deleting the temporary file once it's saved to the DB. I'm going to stop that and then compare the temporary file with the version in the database to see if theres a difference.
15:30.45*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
15:33.41SamotWell...
15:33.55SamotYou say it worked on v11 but it gave warnings...
15:34.57*** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-cucmduklxybntkmn)
15:35.54dan_jThe voicemail system playback gives warnings (and doesnt play). Not the recording.
15:36.09dan_jBut only if i've used my agi script to save the recording to the database
15:36.32SamotBecause VoiceMailMain() is doing a lot of the work.
15:36.39SamotIt records the file in .raw format.
15:36.43SamotThe converts it.
15:37.21dan_jI've downloaded the blob that's created by VoiceMailMain() and it's the same format as the blob generated by my agi script.
15:37.53SamotSo on v11 it doesnt work.
15:38.00SamotJust gives warnings..
15:38.06SamotOn v13 it doesn't work..
15:38.18SamotBut those warnings have turned into errors..
15:38.32SamotEither version VoiceMailMain() has no issues.
15:38.46dan_jAlmost correct. It works fine on my old v11 boxes.
15:39.01SamotSo it doesn't give warnings?
15:39.04dan_jand it's been working for months. But i'm moving to v13 where it doesnt work. So I set up a test v11 box and it doesnt work on that either.
15:39.08igcewielingdan_j: I've not read all the scrollback.  Any no chance your AGI is ending early because the caller hung up?
15:39.35dan_jNo warnings on the old v11 boxes where it works. Only the new v11 boxes.
15:39.47SamotWhat version of v11 are you running that it works on?
15:39.53igcewielingdan_j: check the charset of the tables, maybe a show create.
15:39.57dan_jigcewieling: Nope. It's me doing the calling and the confirmation message only gets played once its saved to the db
15:40.44SamotWhat version of v11 are you running that it works on?
15:40.44dan_jigcewieling: just going to compare the original file to the file that's retrieved from the db to see if they match. However the DB schema was taken from the old db so the charset should be the same. I'll check that now.
15:40.46igcewielingalso check mysql versions, that has bitten me in the past.
15:41.08dan_jigcewieling: Just upgraded so the db version matches the working one.
15:41.12SamotWhat version of v11 are you running that it works on?
15:41.27dan_j11.10.2
15:41.36SamotThank you.
15:41.39dan_jThats the version I put on my test box (which doesnt work)
15:41.52igcewielingMysql has a setting to log queries, that has helped me in the past too.
15:42.05dan_js/put/I also put
15:42.16SamotVoilMailMain() stores messages in the DB without issue.
15:42.16dan_jhmm. how do you use that :)
15:42.22dan_jyep
15:42.26Samoter VoiceMailMain()
15:42.46SamotThe only time the issue happens is when he records the greetings in his own way.
15:43.04SamotI don't think the database has anything to do with this.
15:43.11SamotI think he's recording greetings wrong.
15:43.23SamotStored in the DB or straight on the system.
15:43.37dan_jI dont get it. The temporary file and the file that I extract from the DB are identical. Asterisk just doesnt want to play it.
15:44.16igcewielingdan_j: any chance the problem file is stereo, not mono?
15:44.28dan_jThe temporary file can be played, but the identical file from the DB causes Voicemail() to either warn in v11 or deadlock in v13
15:44.46SamotI said format issue.
15:44.49SamotIt was dismissed.
15:45.01SamotThis is why the exercise happened.
15:45.03*** join/#asterisk rmudgett (rmudgett@nat/digium/x-msqonwtntbqpwwyk)
15:45.03*** mode/#asterisk [+o rmudgett] by ChanServ
15:45.08dan_jigcewieling: It's recorded from within asterisk and asterisk can play it back prior to saving to the db
15:45.12SamotTo show it most like _is_ a format issue.
15:45.29Samotdan_j: Youre using a different app for that
15:45.45SamotYou're comparing Playback to how Voicemail() runs.
15:46.02SamotYou have no idea what VoiceMailMain() or Voicemail() are doing.
15:46.12SamotAre they converting the file?
15:46.18dan_jI dont understand how it can be a format issue if the format saved by VoicemailMain is identical to the format saved by my agi script.
15:46.31SamotWhat format is that?
15:46.35igcewielingdan_j: how are you determineing "sameness"
15:46.36SamotWhat is the exact format?
15:47.29dan_jigcewieling: I took the temporary file from before it's inserted into the DB, i used mysql workbench to save the blob to a 2nd file and I used a diff program to compare the two files.
15:47.55SamotNo.
15:48.01SamotWhat audio format settings?!
15:48.09dan_jyes. im getting that
15:48.30igcewielingmd5sum might be better for that, but I doubt that makes a difference
15:48.39dan_jthe one from the db is 8000Hz Mono 16bit Waveform Audio
15:49.41SamotThe one your AGI created?
15:49.48dan_jsame
15:50.03SamotAnd the one VoiceMailMain created?
15:50.48dan_jAll the same. According to audacity
15:50.51*** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright)
15:50.51*** mode/#asterisk [+o seanbright] by ChanServ
15:51.43dan_jgoing to check the charset to see if mysql to sending the info back to asterisk differently.
15:53.25dan_jNope. both charsets are the same between the working server and the other servers.
15:53.37dan_jI'm going to double check that the working server is actually still working.
15:57.11*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
15:58.15dan_jYep. Still works perfectly on the old v11 box. I'm confused.
16:07.35dan_jWeird. I commented out ;format=wav49|gsm|wav and it's working now. 20 hours later
16:09.37SamotI bet format=wav would work too.
16:09.53SamotThat was the other thing, VoiceMailMain() was making three versions of the file.
16:09.59SamotAnd using all three..
16:10.10SamotYour AGI created one format of three.
16:12.01dan_jVoiceMailMain stores all 3 files in the blob field?
16:12.48SamotNo.
16:13.11SamotOr maybe. I really don't know.
16:13.23SamotOr it tries to use those formats to convert the file to best format.
16:13.28dan_jJust looked at my v13 box and it already had format=wav, and thats the one that deadlocks. :(
16:13.58dan_jThis evening, I'm going to comment out format=wav, so there is no format line and see if it still deadlocks.
16:14.12*** join/#asterisk miralin (~Thunderbi@194.8.128.48)
16:29.17*** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1)
16:32.22*** join/#asterisk luckman212 (~luckman21@gateway/vpn/privateinternetaccess/luckman212)
16:41.25*** join/#asterisk Centinel (~jsmith@fsf/member/Centinel)
16:42.30CentinelWe're running an Asterisk 13.13.1 PBX. We've got a bunch of Cisco 7940s set up to register to this PBX. I can see the REGISTER messages coming in through the Asterisk CLI, but it looks like Asterisk simply isn't processing them. Any ideas?
16:43.39*** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0)
16:43.39*** mode/#asterisk [+o DivideBy0] by ChanServ
16:45.04SamotCentinel: So Asterisk isn't receiving the REGISTER and sending back a 401 Unauthorized challenge?
16:45.37*** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n)
16:45.37*** mode/#asterisk [+o cresl1n] by ChanServ
16:46.44CentinelNo - I can see Asterisk receiving the REGISTER messages in the CLI. I used sip set debug to take a closer look at one of the phones. It's just that nothing appears to be happening in response.
16:47.15CentinelWe have some newer Cisco phones - SPA525G - and they can register just fine.
16:47.50SamotAre our 79xx phones using the SIP firmware?
16:48.06SamotCisco 79xx phones are designed for the Cisco UC platform.
16:48.17SamotThey have to be updated/flashed to the SIP firmware.
16:48.26SamotIn order to do SIP.
16:48.41SamotCisco SPA 5xx phones are SIP phones purely.
16:49.06CentinelI assume so. If I change host=dynamic to host=phone_ip_address, calls work just fine.
16:49.37SamotBecause it's not registering.
16:49.38CentinelThere was a very major change yesterday, though. We had been running an ancient copy of Asterisk 1.4. A power surge blew out the old server it was running on, so we had to switch over to an Asterisk 13 VM I had been setting up for my (eventual) transition to Asterisk administration.
16:49.42SamotIt's just sending calls to that IP
16:49.53CentinelExactly.
16:50.01SamotSo it's not registering.
16:50.10CentinelBut only on the old phones.
16:50.57SamotShow the register attempts.
16:51.13SamotWith the host=dynamic for it.
16:51.16SamotNot the IP.
16:51.53Centinelhttp://pastebin.com/Bj3DvQzM
16:52.58SamotSo the settings for this device from sip.conf
16:53.02SamotMask the secret.
16:55.12Centinelhttp://pastebin.com/uPLhPrP7
16:56.49*** join/#asterisk bradley1 (~bradley@207.58.228.18)
16:57.48Centinel@bradley1 is working on this too, @Samot.
16:58.09SamotWhere's the secret?
16:58.21CentinelThere isn't one for this particular device.
16:58.37SamotThen how will a register work?
16:58.57SamotTheres nothing for it to build the digest challenge with.
16:59.09SamotOr the check against the challenge.
17:00.16bradley1I don't think a secret is required?
17:00.16CentinelThere are a few of the new phones that don't have a set secret either. They seem to register okay.
17:00.40SamotAsterisk is clearly receiving the REGISTER
17:00.50SamotIt's not sending back a 401 challenge.
17:03.06CentinelWell if that's what will fix it, I can certainly think of worse things.
17:03.37SamotSet up the device config with the secret and see if the 401 is sent.
17:08.49igcewielingI'm starting to think the best way to provide T.38 fax service to our customers is to simply outsource it.
17:09.57SamotWhys that?
17:10.07SamotI don't have issues with it.
17:10.24igcewielingfailing for no reason is the most common.
17:10.43igcewielingNot even involving Asterisk.
17:11.26igcewielingI assume it is some weird interaction between the MSX/Nextone and asterisk and/or Adtran endpoints.
17:11.58SamotHrmmm.
17:12.05SamotI've done it over Adtrans will little issue.
17:12.21igcewielingEvery few months I try routing it via Asterisk instead of direct to the endpoint but it never woks.
17:12.25SamotIncluding sending it to the Adtran and breaking it into FXS or PRI.
17:12.27igcewielingSamot: how many adtrans?
17:12.36SamotWell this was when I was at the CLEC.
17:12.37igcewielingwe have 100+ adtrans with faxing.
17:12.47SamotSo all of them.
17:13.01SamotOh, way more than that.
17:13.15SamotWe covered all of Michigan.
17:13.15igcewielingOh, I realize it is SUPPOSED to work.  It doesn't for us.
17:13.24SamotParts of WI, IL, IN
17:13.33igcewielingLike I said, I assume it is an issue with the Nextone/MSX "SBC"
17:13.40SamotThat is possible.
17:15.23igcewielingour current setup is:  Level 3 -> our MSX/Nextone SBC -> Adtran endpoint.   and adtran endpoint -> kamailio proxy to add PAID -> MSX/Nextone -> level 3.
17:15.39igcewielingfirst is inbound, second is outbound.
17:15.46Centinel@Samot Added a secret to the phone. Asterisk still picks up the REGISTER messages and still doesn't act on them.
17:16.49SamotShow it.
17:17.11SamotWell I don't know about Level 3
17:17.19SamotBut at the CLEC we had our own network.
17:17.23Centinelhttp://pastebin.com/yHW12wcv
17:17.51CentinelThe only change in the sip.conf context is the addition of a secret. Everything else is the same.
17:21.33SamotOne sec.
17:24.58SamotCentinel: Do: sip show peer 0807
17:25.06Samotpb the output.
17:26.26Centinelhttp://pastebin.com/xGkn2RBP
17:27.45SamotCSeq: 104 REGISTER <-- That's not right.
17:28.33SamotCentinel: Keep the SIP debug going for that device...
17:28.38CentinelThe number started at 100 and has been incrementing with each new set of registration messages.
17:28.47SamotPower down the device..
17:28.52SamotLet it sit for a sec..
17:28.55SamotPower it back up..
17:29.00SamotShow the register from that.
17:29.23SamotCSeq: 860 REGISTER <-- That was the first debug. That's still wrong.
17:29.26SamotIt needs to be a 1
17:30.43CentinelI'm wondering if the version of the firmware on the phone might need updated now that we've jumped a whole bunch of Asterisk versions. But yes, I power cycled the device.
17:30.47CentinelWaiting for output now.
17:31.13SamotIMHO: 79xx phones should be introduced to sledgehammers.
17:32.14CentinelOnly if I get to do it Office Space-style.
17:33.37SamotThat's how we did it.
17:33.52SamotWhen the CLEC finally replaced those shitty ass phones.
17:35.14CentinelThe first output has arrived, and it looks like it's back to 103: http://pastebin.com/kzHdRX7H
17:37.00CentinelAnd incrementing again.
17:37.35CentinelIf nothing else, this is certainly an interesting introduction to IP telephony. I hadn't planned to dive into this for another few months.
17:38.02CentinelAlthough I'm not entirely clueless. I'm in the middle of one of Digium's training courses.
17:46.47SamotPhones and devices are local to the PBX
17:46.48Samot?
17:47.32CentinelAs in, on the same subnet and VLAN? Yes.
17:48.24Samotpermit=10.0.16.0/255.255.252.0 <- Change that 0.0.0.0/0.0.0.0
17:48.29SamotHumor me on it.
17:48.41SamotThen do a sip reload
17:50.51*** join/#asterisk Tiffon (~name@unaffiliated/tiff0n)
17:52.58CentinelWe're going to take a short break and then set up a dedicated test phone. After that's done, we'll see if that helps.
17:57.26*** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net)
17:59.14*** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net)
18:01.33*** join/#asterisk tuxian (~tuxian@194.12.3.78)
18:03.20*** join/#asterisk Juggie (~Juggie@unaffiliated/juggie)
18:03.45CentinelIt seems a firmware update fixes the problems.
18:05.41SamotOK.
18:24.01*** join/#asterisk skywayskase (~skywayska@67.139.42.219)
18:41.39*** join/#asterisk jkroon (~jkroon@uls-154-73-32-15.wall.uls.co.za)
19:04.16*** join/#asterisk u0m3 (~u0m3@5-12-7-222.residential.rdsnet.ro)
19:05.18miralinhi guys, does anyone experienced in HEP and webrtc? I have some webrtc clients, they connect to my asterisk through websocket and also i have HOMER that recieve from asterisk call information through HEP protocol. The problem is that for my webrtc clients in source and destination field asterisk paste the same ip (client ip) and in call flow it's looks like client send himself all invite, register and other sip packets. Any idea?
19:05.55*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
19:08.29dan_jWhere can I find the database schema for musiconhold realtime?
19:16.10*** join/#asterisk skywayskase (~skywayska@67.139.42.219)
19:17.53dan_jFound it
19:22.54*** join/#asterisk skywayskase (~skywayska@67.139.42.219)
19:24.59*** join/#asterisk libardi (~libardi@187.64.235.241)
19:46.05dan_jIs it possible to reload extconfig.conf without a restart?
20:04.58*** join/#asterisk anpi (~crr@unaffiliated/anpi)
20:13.45*** join/#asterisk skywayskase (~skywayska@67.139.42.219)
20:18.25*** join/#asterisk skywayskase (~skywayska@67.139.42.219)
20:23.31cuscotry config reload /path/to/conf
20:23.37cusco(in asterisk cli)
20:24.04dan_jthanks. that seems to have done it
20:24.48dan_jhmm. maybe not. it's not added the new moh classes from the db. Never mind. I'll just have to wait. Thanks anyway
20:26.23*** join/#asterisk ChannelZ (channelz@burner.com)
20:35.33*** join/#asterisk agent_white (~agent_whi@unaffiliated/agent-white/x-6197888)
20:36.19cuscodan_j: also reload musiconhold.conf ?
20:36.23cuscoor reload the moh module
20:41.52*** join/#asterisk imcdona (~imcdona@2607:f0d8:20:1001:94fe:2a00:16b0:7ff2)
20:42.34*** join/#asterisk dmasiero (~doug@lantern.masiero.us)
20:45.38*** join/#asterisk skywayskase (~skywayska@67.139.42.219)
20:50.32*** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net)
20:52.59dan_jSamot: It appears that the v13 deadlock is a separate issue. I've commented out ;format=wav and it still deadlocks when trying to playback a file created by my agi script.
20:53.15dan_jGoing to upgrade to v13 on my test box and see if the same occurs.
20:55.09*** join/#asterisk netman (~netman@185.94.249.77)
21:03.10*** part/#asterisk kharwell (kharwell@nat/digium/x-fqssxtindnnaycbp)
21:15.46*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
21:18.08*** join/#asterisk skywayskase (~skywayska@67.139.42.219)
21:20.08dan_jHi. I've got an AGI script with allows a user to record their voicemail greeting outside of app_voicemail. in v11 it works fine. But in v13, Voicemail() is unable to play the greeting recorded by my AGI script. The recordings are stored in a realtime db.
21:20.22dan_jAny ideas how I can diagnose this? I've been fighting with it for over 24 hours now.
21:20.30dan_jAny help is greatly appreciated!
21:22.52dan_jI've just confirmed the issue by downgrading to v11 and accessing the same voicemail box without re-recording. In v11 it plays fine. In v13, the channel deadlocks.
21:26.23*** join/#asterisk rmudgett (rmudgett@nat/digium/x-whukezsdldptztca)
21:26.23*** mode/#asterisk [+o rmudgett] by ChanServ
22:08.13*** join/#asterisk rwb (~Thunderbi@65.183.151.239)
22:10.37*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
22:17.12dan_jIf I extract the recording from the database and play it using Playback(), asterisk is able to play it.
22:19.55dan_jEven weirder, if I take that message and insert it as a voicemail message, it plays fine too. It just causes a deadlock when it's a greeting.
22:37.55*** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net)
22:50.04*** join/#asterisk kharwell (~kharwell@user-24-214-15-130.knology.net)
22:50.04*** mode/#asterisk [+o kharwell] by ChanServ
22:59.39*** join/#asterisk monsterco (~monsterco@toroon474aw-lp130-02-70-50-209-133.dsl.bell.ca)
23:00.40*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
23:00.40*** part/#asterisk monsterco (~monsterco@toroon474aw-lp130-02-70-50-209-133.dsl.bell.ca)
23:04.42*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
23:55.27*** join/#asterisk jeffspeff (~Jeff@209.141.208.197)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.