| 00:01.39 | WIMPy | amonk: That sounds like a bit of hit or miss to me. |
| 00:08.34 | amonk | WIMPy: yeah, it's not ideal, but it looks doable |
| 00:09.23 | WIMPy | Using somethign end2end through something that by definition is an endpoint soulnds like a questionable concept anyway. |
| 00:09.53 | WIMPy | You should look a SIP proxys instead of an B2BUA for that. |
| 00:10.48 | amonk | yeah, i know. |
| 00:13.52 | MaliutaLap | WIMPy: http://www.cisco.com/image/gif/paws/113048/reg-9971ip-cucm.pdf <- seems to indicate DN is "Directory Number" |
| 00:14.34 | WIMPy | MaliutaLap: But where would that appear in any of the XML files? |
| 00:15.19 | WIMPy | I don't find anything number other that the voicemail extension. |
| 00:24.50 | MaliutaLap | WIMPy: yeah, not finding much on getting the 9971's working in that way |
| 00:25.02 | MaliutaLap | WIMPy: most example are just to a single * server |
| 00:25.40 | WIMPy | Guess that's just a case of bad luck. |
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| 02:11.19 | sweettea | ive asked this before, but I am sorry to ask again. I am switching asterisk servers shortly. How can I transfers voicemail? Ive already configured it in the new host, can I just copy the contents in NEW to the new box? |
| 02:12.03 | sweettea | INBOX rather... |
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| 02:54.57 | ChannelZ | copy the whole mailbox dir (generally /var/spool/asterisk/voicemail/[context]/[mailbox] |
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| 03:43.34 | wyoung | sweettea: rsync works great |
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| 07:12.10 | BillyCrook | I'm looking for an IAX provider that I can port my t-mobile number to, and use that number for termination/origination of voice, SMS, and MMS |
| 07:12.21 | BillyCrook | any recommendations? |
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| 09:10.04 | DonkeyDong | Has anyone ever experienced SNOM 3xx phones suddenly starting ignoring incoming traffic? |
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| 10:16.57 | drmessano | BillyCrook: IAX? Don't do it |
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| 12:07.48 | apavlov | Hey all. Wondering here if it's possible to set up a tlsbindaddr with an IPv4+IPv6 interface/port in http.conf (using 13.6). tlsbindaddr=[::]:8089 apparently does not work |
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| 12:39.27 | pawiecki | DonkeyDong: Can you elaborate? |
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| 12:59.17 | DonkeyDong | pawiecki: phones starts ignoring incoming traffic out of the blue. Wireshark shows traffic both ways, ifconfig shows rx packets increments but the phone does nothing with it so it can not communicate with the network. It acts as if it does not see the incoming packets |
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| 13:00.08 | WIMPy | ifconfig on the phone? |
| 13:00.23 | DonkeyDong | Yes |
| 13:00.35 | WIMPy | How do you get there? |
| 13:00.44 | DonkeyDong | serial cable |
| 13:01.04 | WIMPy | Sounds desperate :-) |
| 13:01.20 | WIMPy | Only sip or all traffic? |
| 13:01.29 | DonkeyDong | hehe, all traffic.. |
| 13:02.40 | WIMPy | I havent' seen or heard of anythign the like so far. |
| 13:02.40 | pawiecki | DonkeyDong: latest firmware? I had some problems with firmware on SNOM 370 and 710 and had to downgrade. Another thing is, is the phone registered? Check another phone, with the same settings/firmware combo. |
| 13:03.18 | WIMPy | But yesterday I took apart a Snom for the first time. Don't know why. Usually I have to take a peek earlier :-) |
| 13:03.51 | pawiecki | Just out of curiosity - what phones have you guys used/configured/repaired so far? |
| 13:04.08 | WIMPy | Yeah, I stopped at 8.4.35 as well. |
| 13:04.09 | pawiecki | I have one Polycom VVX500 right now as a organ donor |
| 13:04.24 | pawiecki | repair of the screen costs more that new unit |
| 13:05.23 | WIMPy | Allnet, Avaya, Cisco, Digium, Gigaset, Linksys, Snom |
| 13:07.28 | WIMPy | Siemens, UTStarcom and ZyZEL. |
| 13:08.43 | pawiecki | I had so far: Polycom, RTX, Yealink, Gigaset, Linksys, Cisco, Grandstream, Ascom and Snom. |
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| 13:22.11 | eschmidbauer | we have asterisk setup with public + private NICs |
| 13:22.38 | eschmidbauer | user sends INVITE on private NIC, asterisk sends b-leg INVITE on public, but the to-header has the private NIC IP |
| 13:22.50 | eschmidbauer | is that normal? or should it be translated to public IP |
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| 13:23.16 | WIMPy | Sounds like you haven't configured NAT support correctly. |
| 13:23.56 | WIMPy | With the phones using what they found to be their public IP and Asterist believing them. |
| 13:24.49 | eschmidbauer | i read RFC and it said the to-header contains the logical address. |
| 13:25.17 | eschmidbauer | my question is, does this matter? call seems to setup properly and all. i'm just not sure if it's correct behavior |
| 13:27.32 | WIMPy | Who does that work? I can't see how the phones could be reached vir the public IP. |
| 13:28.35 | eschmidbauer | it's the to-header |
| 13:28.41 | eschmidbauer | only |
| 13:29.00 | eschmidbauer | AFAIK the to-header is not used to navigate SIP packets |
| 13:29.16 | WIMPy | If the phines don't care... |
| 13:29.23 | WIMPy | phones |
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| 14:03.04 | *** join/#asterisk ArcticFox (~arcticfox@95-31-33-189.broadband.corbina.ru) |
| 14:03.10 | ArcticFox | Hello every1! |
| 14:04.31 | ArcticFox | How to fix that error: ERROR[30513]: app_voicemail.c:2813 inboxcount2: Couldn't find mailbox 1001 in context internal. |
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| 14:05.27 | ArcticFox | extension in dialplan presents, in voicemail.conf string also presents: 4001 => 4001,4001 VoiceMail,4001@1ans.net |
| 14:05.59 | WIMPy | How would 1001 match 4001? |
| 14:06.08 | ArcticFox | whooh, yeah found it here |
| 14:06.21 | ArcticFox | 1 moment, fix it in .conf |
| 14:07.05 | ArcticFox | fixed to 1001, same error |
| 14:07.51 | WIMPy | And BTW: Are you still using Asterisk 1.2 or earlier? |
| 14:07.53 | ArcticFox | 1001 => 1001,1001 VoiceMail,1001@1ans.net ; Password is the same as extension |
| 14:08.01 | ArcticFox | Asterisk13, PJSIP |
| 14:08.13 | ArcticFox | not using users.conf |
| 14:08.20 | WIMPy | Oh, that still accepts that syntax? |
| 14:08.30 | WIMPy | Since 1.4 we use () for parameters. |
| 14:09.13 | WIMPy | and "1ans.net" is not "default". So there's still somethign wrong about your story. |
| 14:10.33 | ArcticFox | Oh, thanks, 1 more glitch |
| 14:11.30 | ArcticFox | not that one, creating pastebin |
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| 14:14.37 | ArcticFox | http://pastebin.com/UG9tUAFy |
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| 14:17.01 | ArcticFox | Any ideas? |
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| 14:24.11 | WIMPy | Do you really have [internal] twice? |
| 14:24.39 | WIMPy | BBL |
| 14:25.33 | BillyCrook | drmessano: And why is that? Please elaborate. I thought IAS was the preferred format? |
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| 14:52.56 | WIMPy | BillyCrook: It's not a format. And it's better so people don't like it. |
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| 15:08.19 | qakhan | what is alternate of eventwhencalled and eventmemberstatus in asterisk 13.6 |
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| 15:08.25 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 15:57.38 | *** join/#asterisk onebree (d1bf0df7@gateway/web/cgi-irc/kiwiirc.com/ip.209.191.13.247) |
| 15:58.06 | onebree | hello |
| 16:05.47 | newtonr | onebree, howdy |
| 16:07.09 | onebree | Is it possible to determine what cipher the client-side of things is using? |
| 16:07.43 | onebree | Right now, we believe our issue with video is that the two clients do not have the same cipher/crypto set. |
| 16:08.35 | [TK]D-Fender | Do you have audio? |
| 16:09.05 | [TK]D-Fender | Because if you have one then you can have the other. If you don't have one then the call would have dropped like a rock |
| 16:09.15 | [TK]D-Fender | So basically that isn't your problem |
| 16:09.20 | onebree | The clients cannot hear each other, but they can hear the sounds that asterisk sends |
| 16:09.33 | [TK]D-Fender | Then you are screwing something else up. |
| 16:09.56 | onebree | And that is correct -- when one person enters the conference, all is fine. When a second person enters, the second person is dropped immediately. |
| 16:10.12 | [TK]D-Fender | Show the actual backup for all this |
| 16:10.30 | onebree | This is similar to what I am experiencing, just with sipml5 (not linphone) -- http://forums.asterisk.org/viewtopic.php?f=1&t=93987&start=0&sid=1fb2e14864475e08cf1bc00f9bb6af3a |
| 16:11.49 | onebree | Although closed, this patch seems to not be applied in asterisk for us: https://issues.asterisk.org/jira/browse/ASTERISK-17282 |
| 16:14.13 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 16:15.06 | [TK]D-Fender | Other people's debug is useless garbage |
| 16:15.08 | [TK]D-Fender | Show yours |
| 16:16.26 | onebree | I do not have much -- I keep getting the error "SRTP unprotect failed with: authentication failure" with codes 10 or 110. When the second person enters the conference, the websocket is immediately closed, and a stale connection remains |
| 16:16.33 | [TK]D-Fender | Got GET debug |
| 16:16.41 | [TK]D-Fender | Stop showing us single-line messages |
| 16:16.48 | [TK]D-Fender | that doesn't proce what the comms were that CAUSED it |
| 16:17.03 | [TK]D-Fender | prove* |
| 16:17.18 | onebree | How do I debug or check the client-side ciphers/crypto settings? |
| 16:17.43 | [TK]D-Fender | this is all negotiated in SIP debug. |
| 16:17.48 | *** join/#asterisk Echo6 (~Echo6@64.136.247.50) |
| 16:17.50 | [TK]D-Fender | Stop wasting time and show us the call |
| 16:18.48 | onebree | I don't mean to waste time. I did not know if this could be answered shortly. |
| 16:19.03 | onebree | I'll gather some debugs and if I still need help, I will report back. |
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| 16:35.20 | onebree | I mean, how much of the debug do you want? There is a lot, and I am not sure which would be relevant to show |
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| 16:44.56 | [TK]D-Fender | Stop stalling and start showingh |
| 16:47.14 | onebree | IDK what to show for the debug. A ton of logs just spew out with only one client connected to the conference room |
| 16:47.26 | onebree | I do not have the ability to have 2 clients in the room at once, ATM |
| 17:03.21 | newtonr | [TK]D-Fender, probably easier to specify what you want to see rather than simply saying "show us the call" over and over. |
| 17:03.28 | newtonr | onebree, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 17:03.52 | newtonr | probably want to include "sip set debug on" or "pjsip set logger on" depending on what you use |
| 17:04.54 | newtonr | That will show the SIP packets received and sent by Asterisk along with the debug the rest of the article has you enable. |
| 17:05.00 | [TK]D-Fender | Has has debug and is looking for an excuse to show LESS of it |
| 17:05.33 | *** join/#asterisk ArcticFox (~arcticfox@95-31-33-189.broadband.corbina.ru) |
| 17:05.39 | newtonr | I think that is an assumption you have made. He appears to not understand what you want to see or how to gather it. |
| 17:05.48 | newtonr | He even said "IDK what to show for the debug" |
| 17:05.52 | [TK]D-Fender | "There is a lot" <- why don't see irt NOW? Why are you truying to show LESS? Why the DELAY? |
| 17:05.57 | [TK]D-Fender | He HAS it according to that |
| 17:06.25 | [TK]D-Fender | "There is a lot". So where is it? |
| 17:06.30 | [TK]D-Fender | PAstbin |
| 17:06.32 | [TK]D-Fender | ALL OF IT |
| 17:06.38 | [TK]D-Fender | Let us soprt |
| 17:06.47 | [TK]D-Fender | let us sort through it |
| 17:10.33 | newtonr | Whether he has it or not, typing in caps (which most interpret as yelling) and asking the same thing over and over when the other person obviously doesn't understand doesn't tell anyone how to provide the debug or gather it accurately. If anything it simply makes people irritated. :) |
| 17:11.11 | newtonr | If you believe they do understand but are being obstinate then there is no reason to continue dialog with them. |
| 17:15.45 | moe` | classic, [TK]D-Fender looking for complete logs. newtonr, do what [TK]D-Fender asks, he will fix it for you. |
| 17:16.01 | [TK]D-Fender | onebree> IDK what to show for the debug. A ton of logs just spew out with only one client connected to the conference room |
| 17:16.13 | [TK]D-Fender | Yup, has tons. Shows nothing. Clearly obstinate. |
| 17:16.17 | [TK]D-Fender | You're right |
| 17:16.32 | [TK]D-Fender | moves on to more productive matters |
| 17:17.00 | newtonr | moe`, I'm not the one looking for help. :) |
| 17:17.22 | moe` | if you play nice, the greybeards here will help you fix it |
| 17:17.28 | moe` | they did so for me. |
| 17:17.36 | ArcticFox | Hello, what I'm doing wrong? Asterisk13+PJSIP+Voicemail here is a paste bin: http://pastebin.com/2jBfYCLm |
| 17:18.31 | moe` | one thing that got me twisted six ways from Sunday was a SIP trunk not registering but still working. |
| 17:18.40 | moe` | twilio |
| 17:18.55 | [TK]D-Fender | ArcticFox, Interesting.... |
| 17:19.22 | [TK]D-Fender | actCLI seems to show it there pretty clearly in the VM dump, and the config looks like it matches... |
| 17:20.07 | [TK]D-Fender | ArcticFox, Are you using realtime at all? |
| 17:20.16 | ArcticFox | nope, Extensions.conf |
| 17:20.27 | [TK]D-Fender | ArcticFox, Only thing I can think of is if that dump came from one source but * was looking somewhere else. |
| 17:20.29 | *** join/#asterisk C1ph3r4 (~C1ph3r@177.159.131.189) |
| 17:21.02 | [TK]D-Fender | ArcticFox, try restarting * completely |
| 17:21.10 | ArcticFox | may be users.conf, but it's not usable by pjsip, only by voicemail |
| 17:21.38 | ArcticFox | but i don't know algorith by which parameters from users.conf is taken by voicemail |
| 17:21.54 | [TK]D-Fender | ~users.conf |
| 17:21.54 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 17:22.01 | [TK]D-Fender | Stop using that |
| 17:22.03 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:22.21 | moe` | LOL [TK]D-Fender |
| 17:22.25 | ArcticFox | yeah, this is reason why i don't using it |
| 17:22.27 | moe` | how good is that |
| 17:22.41 | ArcticFox | idea! |
| 17:22.42 | [TK]D-Fender | trash it |
| 17:23.01 | ArcticFox | yep, this is my idea ;) |
| 17:27.14 | moe` | hey, [TK]D-Fender, dialplan can do logic like select lowest latency SIP trunk to select outbound, right? |
| 17:27.52 | moe` | I have 3 SIP trunks, so I'd like to enable automagic failover, etc. |
| 17:28.41 | WIMPy | moe`: The latency is not available. |
| 17:28.43 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 17:29.54 | moe` | WIMPy: dammit, so how to select? |
| 17:30.40 | moe` | right now I have dial prefixes... 9 = flowroute, 7 = twilio, 5 = skype SIP |
| 17:30.43 | moe` | that's LAME |
| 17:30.48 | WIMPy | By destination (cost)? |
| 17:30.56 | moe` | I care not of cost |
| 17:31.06 | moe` | mostly just need to make sure the call completes |
| 17:32.34 | moe` | so it flowroute is down, for example (that's never happened flowroute is awesome).... but if a user tries 9-international, it should failover to twilio or skype |
| 17:32.52 | moe` | s/so it flowroute/so if flowroute/ |
| 17:33.09 | moe` | lol I just sparked infobot, amusing |
| 17:33.48 | moe` | nice work on the bot, what's its running? |
| 17:34.13 | WIMPy | Either chack status before trying with the SIPPEER function, or try and check DIALSTATUS and HANGUPCAUSE and retry on another account when appropriate. |
| 17:34.56 | moe` | ah, ok, so adjust the outbound dialplan and on DIALSTATUS or HANGUPCAUSE to roll over to another provider |
| 17:34.58 | moe` | got ya |
| 17:37.07 | ArcticFox | got interested debug string: [Dec 29 20:33:15] DEBUG[32355] res_pjsip_pubsub.c: Subscription to resource 4001 is not to a list |
| 17:37.07 | ArcticFox | [Dec 29 20:33:15] NOTICE[32355] res_pjsip_mwi.c: AOR 4001 has no configured mailboxes. MWI subscription failed |
| 17:37.07 | ArcticFox | [Dec 29 20:33:15] DEBUG[32355] pjsip: endpoint .Response msg 404/SUBSCRIBE/cseq=2 (tdta0x1fdb320) created |
| 17:38.40 | moe` | Can it be done to setup a virtual SIP? |
| 17:38.58 | moe` | i.e. a virtual SIP trunk that is a mongrel of real trunks |
| 17:39.12 | WIMPy | Isn't it always virtual? |
| 17:39.18 | moe` | well, true |
| 17:39.39 | moe` | but can I define a SIP trunk to be something of a combo of real trunks? |
| 17:39.48 | WIMPy | Nope. have to use dialplan. |
| 17:39.50 | moe` | for example, a simple extension is a virtual, yes |
| 17:40.07 | moe` | that's unfortunate |
| 17:43.09 | [TK]D-Fender | ArcticFox, that message shouldn't be related to your previous error. It is its own issue |
| 17:43.25 | [TK]D-Fender | ArcticFox, fix the endpoint to have that box |
| 17:44.05 | ArcticFox | trashed users.conf, need to add [endpoint] type=aor, mailboxes=vm4001@internal? |
| 17:45.59 | [TK]D-Fender | read the sample config to see wher it belongs |
| 17:46.13 | ArcticFox | no samples with aor&&mailboxes |
| 17:49.15 | ArcticFox | great: [Dec 29 20:48:30] DEBUG[32483][C-00000002] app_voicemail.c: Before find_user |
| 17:49.16 | ArcticFox | [Dec 29 20:48:30] WARNING[32483][C-00000002] app_voicemail.c: No entry in voicemail config file for '4001' |
| 17:54.33 | [TK]D-Fender | ArcticFox, the sample config clearly shows where mailboxes go |
| 17:54.36 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.29.196) |
| 17:55.24 | ArcticFox | are we about: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip ? |
| 17:55.50 | [TK]D-Fender | Is that the sample config? |
| 17:55.54 | [TK]D-Fender | Looks like a wiki page. |
| 17:56.16 | [TK]D-Fender | Go read the sample config |
| 17:57.43 | *** part/#asterisk DragonAzul (~DragonAzu@187.208.29.196) |
| 17:57.58 | ArcticFox | but does mwi required? it conflicts with app_voicemail |
| 17:58.06 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 17:58.28 | [TK]D-Fender | I already told you it' |
| 17:58.32 | [TK]D-Fender | s not related |
| 17:58.39 | [TK]D-Fender | [TK]D-Fender> ArcticFox, that message shouldn't be related to your previous error. It is its own issue |
| 17:59.20 | ArcticFox | k, cleaning configs |
| 17:59.30 | moe` | hey, suppose a guy wanted to plug in an analog phone line to a x86/x64 boxen via PCIe, what's the go there? |
| 17:59.55 | moe` | for asterisk integration, of course |
| 18:00.22 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 18:00.26 | moe` | suggestions? |
| 18:00.39 | moe` | there are many cards out there, anyone with experience on that? |
| 18:00.53 | moe` | it should be a simple thing, really |
| 18:01.20 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 18:02.54 | moe` | so far all I've got is Bria/CSipSimple VoIP for clients, and SIP trunks, but I'd like to extend to actual physical land lines... inviting landlines to the party. |
| 18:03.08 | moe` | well, linphone too, of course |
| 18:03.47 | moe` | so outbound to landlines I have, but I'd like to have a means for inbound landlines - analog or digital. |
| 18:06.04 | sweettea | does the voicemail directory need to be owned by asterisk? |
| 18:06.16 | moe` | I have a DISA 1.888 number, etc, so landlines can call the system and dialout via that |
| 18:06.22 | moe` | but I'd like something more direct |
| 18:06.40 | [TK]D-Fender | It clearly needs rights to all the files |
| 18:06.55 | [TK]D-Fender | Why shouldn't it be the owner? |
| 18:07.05 | [TK]D-Fender | If other things need rights then give it to them |
| 18:07.14 | [TK]D-Fender | Other things shouldn't be the owner of things * makes |
| 18:07.14 | moe` | asterisk daemon needs to have read/write access to the voicemail repository |
| 18:07.39 | [TK]D-Fender | I don't buy a car, say it belongs to my dog, and then say the dog gave me rights to the car. |
| 18:08.35 | sweettea | thats one way to answer it |
| 18:08.36 | sweettea | thanks |
| 18:08.39 | ArcticFox | at last i have a new error ;) good news may be |
| 18:10.42 | sweettea | /var/lib/asterisk is not owned by root, by default, why i asked. Installed put a bunch of stuff in there. |
| 18:11.22 | [TK]D-Fender | it shouldn't be owned by root |
| 18:11.28 | [TK]D-Fender | Since Asterisk shouldn't be run as root |
| 18:12.46 | onebree | Does that mean doing `sudo asterisk -rvvv` is wrong? |
| 18:14.05 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 18:14.39 | sweettea | ahh there lies my problem |
| 18:14.43 | sweettea | its runnin as root |
| 18:14.59 | ArcticFox | new error: http://pastebin.com/HP7gssN9 |
| 18:15.07 | [TK]D-Fender | <onebree> Does that mean doing `sudo asterisk -rvvv` is wrong? <- no |
| 18:15.24 | [TK]D-Fender | that only CONNECTS to an already running * |
| 18:15.25 | moe` | root@banshee:/usr/local/etc/asterisk:0# ps axO ruser | grep asterisk |
| 18:15.25 | moe` | 77873 asterisk - Is 122:14.83 /usr/local/sbin/asterisk -n -F -U asterisk |
| 18:15.30 | moe` | that sorta thing? |
| 18:15.49 | onebree | [TK]D-Fender: Thank you for the clarification. |
| 18:16.12 | [TK]D-Fender | ArcticFox, PS your voicemail.conf |
| 18:16.14 | [TK]D-Fender | PB |
| 18:16.56 | [TK]D-Fender | And show us the contents of your config folder as a whole |
| 18:17.11 | [TK]D-Fender | "ls -la /etc/asterisk" <- or wherever you have them |
| 18:18.59 | sweettea | why does default install run as root? |
| 18:21.07 | [TK]D-Fender | There is no such thing as "default" |
| 18:21.13 | [TK]D-Fender | There is only "how you did it" |
| 18:21.27 | sweettea | i did w/e the wiki said :) |
| 18:21.44 | ArcticFox | http://pastebin.com/8wWhELCT |
| 18:22.17 | [TK]D-Fender | Wiki shows all sorts of creating a user for it.... |
| 18:25.13 | ArcticFox | wiki shows clearly, but something goes wrong |
| 18:25.23 | ArcticFox | and it's in voicemail.conf now |
| 18:25.32 | moe` | [TK]D-Fender any suggestions on a card for analog/digital landline to an asterisk instance via PCI/PCIe? |
| 18:25.35 | ArcticFox | pjsip have no errors now |
| 18:25.46 | [TK]D-Fender | ls /etc/asterisk |
| 18:25.49 | [TK]D-Fender | -la <------- |
| 18:26.51 | ArcticFox | thank you, found new glitch |
| 18:30.56 | ArcticFox | http://pastebin.com/iK7FpnAY |
| 18:34.08 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 18:34.42 | ArcticFox | http://pastebin.com/HMMagiWZ |
| 18:35.03 | ArcticFox | strange rtp source comes, but mostly it's norma; |
| 18:37.14 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:37.14 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:48.53 | *** join/#asterisk millsu2 (~millsu2@69-195-66-124.unifiedlayer.com) |
| 18:49.38 | millsu2 | Is it possible to keep a callers wait time when moving a call to another queue? |
| 18:51.00 | millsu2 | I don't think setting the position will hep, since the members answering the call are members of other queues. |
| 18:52.26 | *** part/#asterisk Ellenor (~x@unaffiliated/ellenor) |
| 18:55.41 | [TK]D-Fender | No, wit time is wait time |
| 18:55.49 | [TK]D-Fender | You can force their position, that is about all |
| 18:56.20 | *** join/#asterisk davlefouAMD (~davlefouA@41.225.206.154) |
| 18:56.40 | ArcticFox | any ideas for me case? |
| 18:59.44 | [TK]D-Fender | set core debug and retest |
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| 19:17.32 | ArcticFox | [Dec 29 22:10:26] DEBUG[985][C-00000003] app_voicemail.c: Before find_user |
| 19:17.32 | ArcticFox | [Dec 29 22:10:26] WARNING[985][C-00000003] app_voicemail.c: No entry in voicemail config file for '4001' |
| 19:24.54 | [TK]D-Fender | gives up |
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| 20:03.17 | ArcticFox | in which context app_voicemail.c looks for mailboxes - mailbox-context(voicemail.conf) or dialplan-context(extensions.conf)? |
| 20:04.27 | onebree | If the context is the mailbox name/number, it looks for it in voicemail.conf. |
| 20:04.53 | onebree | I think of extensions.conf as handling the routing/paths of a call |
| 20:06.27 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 20:15.35 | ArcticFox | than i'm totally lost |
| 20:16.08 | ArcticFox | [internal] |
| 20:16.09 | ArcticFox | 4001 => 4001,VoiceMail vm4001 |
| 20:16.09 | ArcticFox | Correct, is it? |
| 20:16.58 | ArcticFox | for imap string: 4001 => 4001, name, vm4001@test.net |
| 20:17.13 | ArcticFox | pin = extension |
| 20:18.05 | [TK]D-Fender | Why are we talking about imap? |
| 20:18.49 | ArcticFox | I'm using voicemail with imap |
| 20:18.57 | [TK]D-Fender | Why are you only telling us this NOW? |
| 20:19.16 | ArcticFox | I'm thinked that bug was resolved in 2012 |
| 20:19.39 | ArcticFox | does it changes alot? |
| 20:19.41 | [TK]D-Fender | I think you have been holding out on needed information and we have been wasting our time. |
| 20:20.08 | ArcticFox | but imap was in all PBs |
| 20:20.35 | [TK]D-Fender | Where do we see the core debug I told you for? |
| 20:20.41 | [TK]D-Fender | You showed us the text-file config |
| 20:20.55 | [TK]D-Fender | We have no prrof about what voimail module is actualyl even loaded |
| 20:21.02 | ArcticFox | <ArcticFox> [Dec 29 22:10:26] DEBUG[985][C-00000003] app_voicemail.c: Before find_user |
| 20:21.02 | ArcticFox | <ArcticFox> [Dec 29 22:10:26] WARNING[985][C-00000003] app_voicemail.c: No entry in voicemail config file for '4001' |
| 20:21.03 | [TK]D-Fender | and you did not say you were using imap |
| 20:21.19 | [TK]D-Fender | WThat doesn't say it either |
| 20:21.30 | ArcticFox | app_voicemail.so Comedian Mail (Voicemail System) with IM 0 Running core |
| 20:21.47 | [TK]D-Fender | there are MULTIPLE modules |
| 20:21.51 | [TK]D-Fender | Go look |
| 20:21.56 | ArcticFox | omg... |
| 20:22.37 | [TK]D-Fender | You can't load more than one |
| 20:22.44 | [TK]D-Fender | every other one will fail |
| 20:23.59 | ArcticFox | other mails unloaded |
| 20:25.33 | ArcticFox | http://pastebin.com/V8mHU9UW |
| 20:29.44 | [TK]D-Fender | no. |
| 20:29.51 | [TK]D-Fender | go into your MODULES FOLDER and look |
| 20:31.55 | ArcticFox | look for what? |
| 20:39.08 | ArcticFox | cleaned from res_ari_mailboxes and minivm |
| 20:40.35 | ArcticFox | app_voicemail.so |
| 20:40.35 | ArcticFox | res_pjsip_send_to_voicemail.so |
| 20:40.35 | ArcticFox | res_stasis_mailbox.so |
| 20:41.07 | ArcticFox | checked for conflicts in make menu |
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| 21:02.44 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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| 21:06.15 | jpastore | hi have a n00b question. Is there a lexicon that says version 1.8.x = asterisk 1x? because I have no idea what version I have (11,12,13etc?) when I do core show version it returns 1.8.x... |
| 21:06.53 | onebree | Then you have version 1.8.x |
| 21:07.12 | onebree | Asterisk started its versions at 10+ after 1.8 |
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| 21:08.08 | jpastore | ahh...that's a weird jump |
| 21:08.11 | jpastore | so I'm way behind |
| 21:10.45 | onebree | We are still using 1.8 here for production work, but we plan to move to 13.x soon. |
| 21:12.42 | mjordan | ~versions |
| 21:12.43 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 21:12.48 | mjordan | ^^^ |
| 21:14.57 | ArcticFox | how to check app_voicemail configuration? with IMAP |
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| 21:27.42 | jpastore | thanks guys |
| 21:27.45 | jpastore | appreciate the link |
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