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| 01:56.19 | drab | any chance somebody's around that can explain me a bit how MWI is supposed to work? O' |
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| 01:56.30 | drab | I've been trying to get these old phones to work for hrs now to no avail |
| 01:56.50 | drab | they are old, largely undocumented, but the few pdfs I found seem to suggest they should work |
| 01:57.03 | drab | however I get no lights to come up |
| 01:57.38 | drab | even tho I managed to program the key to the voicemail and it works calling in, but the led next to it is always off |
| 01:58.04 | drab | in some places it's like suggesting that you shuold have mwi_from=asterisk in sip.conf, which I added, but no difference |
| 01:58.11 | drab | I also tried the polling, nothing |
| 01:58.22 | drab | should I be seeing something in console with vvvvvvvvvvvvvvvvv about the MWI? |
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| 03:36.16 | socomm | For confbridge, whats the option I should look at to assign an administrator of a conference bridge? |
| 04:21.17 | ChannelZ | drab, if you have a mailbox assigned to the peer it will send a SIP message to the phone when a new message arrives or the status otherwise changes. |
| 04:22.05 | ChannelZ | And/or you can tell the phone it needs to subscribe to asterisk for MWI updates |
| 04:22.57 | ChannelZ | The former is semi-passive, the latter is more active |
| 04:27.00 | drab | ChannelZ: well, that's what I thought I did (former), by setting mwi_from=asterisk, but maybe that's not it? |
| 04:27.24 | drab | also for whtever reason while I was tcpdumping looking for that stuff I saw a whole bunch of SUBSCRIBE/AUTH DENIED |
| 04:27.52 | drab | which I can't explain since the peer successfully registers and works as expected (and if I change the pwd it fails to register and I see that in the logs) |
| 04:28.03 | drab | so I'm not sure what that failure to subscribe is about |
| 04:28.45 | drab | ChannelZ: given that I can receive and listen to my voicemail if I call the extension I assigned to it, would you agree that I have a mailbox assigned to the peer? or does that mean something else? |
| 04:29.07 | drab | I just added an entry in voicemail.conf and it worked |
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| 04:29.25 | drab | this is a stock ubuntu 14.04 with asterisk 11 |
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| 06:24.13 | ChannelZ | drab, no mailbox=123@whatever in sip.conf for the peer is what lets asterisk know which mailbox it should pay attention to for that peer and notify about new VMs |
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| 06:25.13 | ChannelZ | As to your second question, no, being able to get your voicemail isn't related |
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| 07:33.55 | Stevie-O | Hello |
| 07:36.32 | Stevie-O | anybody awake? |
| 07:48.54 | ChannelZ | ish |
| 07:50.11 | Stevie-O | oooh |
| 07:50.13 | Stevie-O | I got it! |
| 07:50.20 | Stevie-O | I had EXTN instead of EXTEN |
| 07:51.30 | Stevie-O | I have successfully created a dial pattern for dialing peoples extensions |
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| 11:35.52 | thiagoc | anyone using the packages from epel centos repo? |
| 11:48.26 | thiagoc | I still have to compile dahdi-linux when using these packages? |
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| 12:45.03 | IPalmer | Hi all, is there anyone here who can help with Realtime using CURL requests instead of a database connection? |
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| 13:25.36 | c0ldg0ld_ | When using DPMA with a D70 what all could get in the way of SmartBLF working? I have the contacts.xml setup as well as the blf_items file setup ??? |
| 13:25.38 | c0ldg0ld_ | <PROTECTED> |
| 13:25.45 | c0ldg0ld_ | oh that's weird |
| 13:26.18 | c0ldg0ld_ | I also have the subscribe_to="auto_hint_SIPUSERNAME" in the contacts file for each entry |
| 13:26.39 | c0ldg0ld_ | but when executing "core show hints |
| 13:27.02 | c0ldg0ld_ | dangit... " it shows no watcher |
| 13:49.23 | Kobaz | So, I have a digium d70, blf config http://pastebin.com/gW5nuKaq and phone contacts snippit http://pastebin.com/wpwYgmnZ |
| 13:49.38 | Kobaz | the phone's not sending any SUBSCRIBEs |
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| 14:01.24 | c0ldg0ld | Kobaz: same problem here |
| 14:01.49 | Kobaz | oh really |
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| 14:02.30 | c0ldg0ld | Kobaz: I'm using DPMA though so I have (according to the wiki) subscribe_to="auto_hint_<sipusername>" |
| 14:04.05 | c0ldg0ld | I've been combing the forums trying to figure out what was wrong, just tried adding a blf_contact_group to the res_digium_phone.conf that matched the contact_group in the contacts xml file |
| 14:04.07 | c0ldg0ld | no joy |
| 14:04.50 | c0ldg0ld | I'm able to see the contacts on the side car, and use the speed dials but not get any device state information, no green "idle" led, no red flashing "ringing" led |
| 14:05.13 | c0ldg0ld | and the hints are showing no watchers when I do "core show hints" |
| 14:05.31 | Kobaz | yeah |
| 14:05.36 | Kobaz | exactly |
| 14:06.02 | c0ldg0ld | I'm mostly certain there's something stupid I'm missing but damned if I can figure out what it is |
| 14:06.23 | c0ldg0ld | the documentation for some of this isn't wonderful yet |
| 14:12.54 | Kobaz | yeah |
| 14:12.56 | Kobaz | well |
| 14:13.00 | Kobaz | i'm gonna plug my vvx back in |
| 14:13.02 | Kobaz | i want my blfs |
| 14:15.27 | c0ldg0ld | ugghhh those damn polycoms are such a pain to provision |
| 14:15.37 | Kobaz | hah |
| 14:15.37 | Kobaz | no |
| 14:15.43 | Kobaz | mine are perfect every time |
| 14:16.05 | c0ldg0ld | I never deployed too many of them so every time I have to work on one i have to re-learn |
| 14:16.14 | Kobaz | i've deployed over a thousand |
| 14:16.41 | Kobaz | i wrote a provisioning server and it builds the xml |
| 14:16.42 | c0ldg0ld | obviously not the phones fault. I've got probably 600 or so OOOOOOLD cisco's in service |
| 14:16.45 | Kobaz | all that stuff is all automated |
| 14:16.51 | c0ldg0ld | moving them over to digium |
| 14:17.02 | c0ldg0ld | yeah I did the same for the 79x0 flat files |
| 14:17.06 | Kobaz | with no blfs apparently, heh |
| 14:17.07 | c0ldg0ld | made it super simple |
| 14:17.17 | c0ldg0ld | well most of the stuff I've deployed are D45's |
| 14:17.28 | c0ldg0ld | first time I've run into the "secretary" thing |
| 14:18.17 | Kobaz | one issue with the vvx is bugs in the blf startup |
| 14:18.18 | c0ldg0ld | Had some SNOM 821's with their "vision" side car in adminstration, those are extremely unreliable |
| 14:18.24 | Kobaz | you have to call each user in order for it to subscribe |
| 14:18.32 | Kobaz | i wonder if that's fixed in a newer firmware |
| 14:18.35 | c0ldg0ld | hmmm |
| 14:18.44 | *** part/#asterisk IPalmer (~IceChat9@88.211.120.74) |
| 14:18.45 | c0ldg0ld | I read about that on the digiums too at one point in this research |
| 14:18.57 | c0ldg0ld | but that's supposed to be fixed now on digium |
| 14:20.36 | c0ldg0ld | Kobaz: Is your provisioning tool just a bash script that reads some stuff into vars then builds the XML? or more complicated? I had some thoughts about building something with a skeleton file and using sed or something to replace known default values with what I read in from the script |
| 14:20.50 | Kobaz | no |
| 14:20.53 | Kobaz | i have a custom filesystem |
| 14:20.54 | Kobaz | using FUSE |
| 14:21.11 | Kobaz | so you can provision on tftp, ftp, https |
| 14:21.15 | Kobaz | using the same files |
| 14:21.33 | Kobaz | it will autoprovision new phones |
| 14:21.53 | c0ldg0ld | So where do you put the configuration information for each phone?\ |
| 14:21.55 | Kobaz | all you need to do is plug them in and assuming it gets right right vlan (lddp/etc) it'll hit the prov server and then create a new extension for itself |
| 14:22.00 | Kobaz | postgres db |
| 14:22.09 | c0ldg0ld | ah |
| 14:22.18 | Kobaz | so if you're plugging in 5 phones or 500 phones, it's like zero work |
| 14:22.23 | c0ldg0ld | that's pretty slick |
| 14:22.28 | Kobaz | you just have someone unbox phones for $10 an hour |
| 14:22.40 | Kobaz | and you dont need to configure anything, unless people want specific exten numbers |
| 14:22.44 | Kobaz | and then i have a script for that |
| 14:22.52 | Kobaz | you dial *9999 and reprogram the phone with the prompts |
| 14:22.59 | Kobaz | "enter new extension followed by pound" |
| 14:23.03 | Kobaz | and then it swaps stuff around in the db |
| 14:23.12 | c0ldg0ld | oh so this assumes no DID and just increments the extension number like a key system unless you specify otheriwse |
| 14:23.21 | Kobaz | yeah pretty much |
| 14:23.25 | Kobaz | dids are easy |
| 14:23.31 | Kobaz | go to our web portal and stick in a did for the extension |
| 14:23.38 | Kobaz | and then it creates the db dialplan |
| 14:23.54 | c0ldg0ld | are you a clec or something? |
| 14:24.08 | Kobaz | a little of everything |
| 14:24.19 | Kobaz | mostly large on-prem call centers |
| 14:24.35 | c0ldg0ld | cool, I'm just pbx admin for a large-ish hospital and it's associated clinics and long term care facility |
| 14:25.06 | c0ldg0ld | out of curiosity, what do you use for parsing CDR records? |
| 14:25.16 | Kobaz | haha |
| 14:25.18 | Kobaz | i don't |
| 14:25.37 | c0ldg0ld | ha, me either, then every once in a while people want information and I have a helluva time getting it for them |
| 14:25.40 | Kobaz | we have a whole analytics platform with our own custom call logging in the db |
| 14:25.47 | Chainsaw | c0ldg0ld: I use a guy called James. |
| 14:25.48 | Kobaz | i don't parse anything |
| 14:25.52 | Kobaz | it's all db queries |
| 14:26.11 | Kobaz | cdrs don't do it for me, so we track at a much more detailed level |
| 14:26.13 | c0ldg0ld | Chainsaw: feel bad for James |
| 14:26.54 | c0ldg0ld | Kobaz: I gotcha, so you're just bypassing that altogether and creating your own. I guess I don't need that kinda granularity. Just basic overview |
| 14:27.11 | c0ldg0ld | Maybe way to make pretty graphs for people to shut them up occasionally |
| 14:27.54 | Kobaz | pretty graphs, try jscharts |
| 14:28.10 | c0ldg0ld | thx |
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| 14:42.55 | c0ldg0ld | Ok, so if I don't use the auto_hints created by DPMA it works |
| 14:43.03 | Kobaz | really |
| 14:43.19 | Kobaz | got some config examples? |
| 14:43.27 | Kobaz | this is my first digium i've programmed |
| 14:44.06 | c0ldg0ld | I built a regular hint and it works. I'm wondering if the fact that I have subscriecontext=hints in my sip.conf file that it's the only place they look for hints |
| 14:44.32 | c0ldg0ld | the auto_hints appear to be in the actual context of the phone not in the hints context... maybe I'll try some includes and see if that helps |
| 14:44.51 | Kobaz | oh, yeah if you dont have a subscribecontext then the sip peer doesn't know where to get the hints from |
| 14:45.32 | c0ldg0ld | well I had one, but it's for my manual hints on the stupid snom phones |
| 14:46.03 | c0ldg0ld | so I can't really change it, but the auto_hints for the DPMA controlled phones show up @somedifferentcontext |
| 14:46.50 | c0ldg0ld | bingo |
| 14:47.09 | c0ldg0ld | include => somedifferentcontext inside the hints context worked |
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| 14:50.37 | EmRe | A reverse-engineering trivia for experts. What causes an Asteriks Server (13.7 & 13.2) to put irrelevant pauses while playing sounds on channel while calling 5-10 concurrent connection but no problem with 1. |
| 14:52.46 | Kobaz | EmRe: you'll need to describe your environment a little more |
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| 14:56.54 | EmRe | Kobaz: I just checked if this is a known issue - I am currently Re-Building All enviroment. And will give it another shot again. If problem presists, i can give full scope of my latest configuration. basicly my last try was on 13.7.2 using chan_sip , ARI via .Net |
| 14:57.16 | Kobaz | i'm more interested in the digium phone config |
| 14:57.35 | Kobaz | c0ldg0ld: ^ |
| 14:58.00 | Kobaz | EmRe: it's hard to check for known issues without specifics |
| 14:58.23 | Kobaz | you didn't describe the techology involved (Are you using a line card, sip, iax, mgcp... webrtc? etc etc etc)? |
| 14:58.52 | Kobaz | is everything chan_sip or just part of it? |
| 14:59.16 | Kobaz | and what applications are you using, are you in a conference bridge? if so, what timing driver are you using? |
| 15:02.41 | EmRe | Kobaz: you are right... just doing chan_sip. nothing fancy. I am not using conference nor bridge. making calls (via .net AsterNet API) , openning channels, doing my on timing via controling start finish events of playing sound (i am not sure if this is what you meant) |
| 15:03.46 | Kobaz | no the timing driver |
| 15:03.52 | Kobaz | are you using timerfd, or dahdi |
| 15:03.54 | Kobaz | or pthread? |
| 15:04.18 | Kobaz | run 'timing test' |
| 15:04.21 | Kobaz | in the asterisk cli |
| 15:06.03 | EmRe | apperantly timerfd |
| 15:06.08 | Kobaz | what kernel version? |
| 15:06.11 | Kobaz | uname -a |
| 15:07.55 | EmRe | currently re-installing all server :( |
| 15:08.05 | Kobaz | that's not going to help, probably |
| 15:08.09 | Kobaz | what's the specs on the server |
| 15:09.14 | c0ldg0ld | Kobaz, will toss up on pastebin in a sec |
| 15:09.38 | c0ldg0ld | again I'm using DPMA though and it sounded like you weren't so it'll be different |
| 15:10.37 | Kobaz | c0ldg0ld: would just be interesting to see, maybe i'm missing somethign in my core config |
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| 15:22.18 | c0ldg0ld | Kobaz: http://pastebin.com/e66JF0RC |
| 15:25.04 | Kobaz | so you have auto hint |
| 15:25.10 | Kobaz | i just have: subscribe_to="sip:5504@192.168.5.2" |
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| 15:30.03 | grim_ | anyone here that works on the asterisk.org wiki???? I found some errors.... |
| 15:30.40 | grim_ | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29394498 |
| 15:31.26 | grim_ | type is NOT app or exten, these options throw errors. proper type is application or extension. |
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| 15:34.16 | file | that is generated automatically, it's kept in the source code |
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| 16:30.11 | c0ldg0ld | Kobaz: yeah the auto_hint has to do with the fact that the phones are controlled by DPMA, if you arenm' |
| 16:30.16 | c0ldg0ld | t using dpma you do it like you have |
| 16:40.24 | Kobaz | yeah |
| 16:40.31 | Kobaz | except it's not subscribing at all |
| 16:46.03 | c0ldg0ld | what do you get when you do core show hints? |
| 16:46.40 | c0ldg0ld | also I just added an "indication" on my blf.xml to make the LED solid red when on a call |
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| 17:38.21 | tsoutseki | hi |
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| 17:46.32 | Windows98SE | hmpf |
| 17:47.03 | c0ldg0ld | that is a fantastic nick |
| 17:47.40 | Windows98SE | course |
| 17:48.25 | c0ldg0ld | downgraded... |
| 17:49.40 | Windows98 | hands c0ldg0ld a windows 98 installation disk |
| 17:51.24 | c0ldg0ld | requests boot floppy as well |
| 17:52.24 | c0ldg0ld | for a long time after I no longer supported any clients on '98 I used to keep that disk around just for fdisk and other utilities on it |
| 17:52.24 | Windows98 | that disk is bootable |
| 17:52.54 | c0ldg0ld | Maybe I was thinking 95... one of them had a CD that you could do nothing with unless you had a floppy with mscdex on it |
| 17:56.49 | tsoutseki | id give you a my floppy booty, c0ldg0ld |
| 17:57.41 | c0ldg0ld | awww |
| 18:00.39 | tsoutseki | c0ldg0ld, <3 |
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| 19:20.37 | drab | ok so I might be making a bit of progress on this MWI issue |
| 19:21.26 | drab | looking at wireshark I can see that the phone is trying to SUBSCRIBE, which accordingly to the internets might be used for MWI, but it gets an AUTHORIZED response twice and then Bad Event |
| 19:21.41 | drab | the thing is, the phone can REGISTER just fine as I can see it in sip show peers and it works just fine |
| 19:22.03 | drab | so I'm not sure why the SUBSCRIBEs are failing |
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| 19:52.36 | pankid | I am trying to have an extension match based on caller id. It works when I set the exact caller id, but when I try to use X or [0-9] it no longer matches. Am I able to use the same pattern matching rules when matching based on caller iD? http://paste.debian.net/381860/ |
| 19:53.14 | [TK]D-Fender | You are failing to specify that what follows is a pattern |
| 19:53.31 | [TK]D-Fender | basic syntax failure |
| 19:53.52 | pankid | oh, so _XXX/_200X ? |
| 19:54.06 | [TK]D-Fender | yes |
| 19:54.22 | pankid | thanks, |
| 19:54.26 | pankid | sorry for the stupid question. |
| 19:55.22 | [TK]D-Fender | Not a stupid question at all, just a silly mistake |
| 19:55.31 | [TK]D-Fender | With the quickest possible solution |
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| 19:59.48 | Windows98SE | astrisk |
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| 21:00.49 | Kobaz | c0ldg0ld: hints show no watchers, because it's not watching... the phone does not subscribe |
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| 21:32.09 | c0ldg0ld | Kobaz: weird, what about the blf_contact_group in your phone configuration xml, does it match the contact_group in your blf xml? |
| 21:32.19 | Kobaz | maybe |
| 21:32.34 | c0ldg0ld | that was one of the two things in my reading that apparently matters a lot |
| 21:32.45 | c0ldg0ld | the other was the subscribe_to |
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| 21:48.38 | RadicalDev | Does the Homer/sipcapture project have an IRC channel? |
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| 21:53.10 | xpheres | hello |
| 21:53.15 | xpheres | I just compiled asterisk from source |
| 21:53.20 | xpheres | everything seems ok |
| 21:53.31 | xpheres | but I can not find configuration files, and etc/asterisk is empty |
| 21:54.06 | xpheres | I installed asterisk 11.21.2 |
| 21:55.59 | WIMPy | 'make samples' |
| 21:56.13 | xpheres | ah ok |
| 21:57.05 | xpheres | now everything is there |
| 21:57.09 | xpheres | another question |
| 21:57.41 | xpheres | I use freepbx at home, but I installed asterisk in my external virtual server because freepbx does not coexist with other web services |
| 21:57.54 | xpheres | does anyone knows an easy way to import configuration from freepbx to asterisk? |
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| 22:14.38 | file | xpheres, you don't/can't |
| 22:14.57 | xpheres | ok |
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| 22:18.59 | klow | Anyone use sngrep ? I built w/ openssl and specified my private key but can't get TLS SIP messages to show only UDP ones . |
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